From jeff.pyle at fidelityvoice.com  Fri Jul  1 04:15:22 2016
From: jeff.pyle at fidelityvoice.com (Jeff Pyle)
Date: Thu, 30 Jun 2016 22:15:22 -0400
Subject: [OpenSIPS-Users] How to append header to stateless reply
Message-ID: <CACYJG3+U4czpLV6c+aiuLg39nw3QuDFpqSZbj4Td6YSpOd90TQ@mail.gmail.com>

Hello,

How can I add a header to a stateless reply?  Or, if that's not possible,
to a transactional reply?

I'm trying to do something like this:

route {
  ...
  append_hf("P-Custom-Header: $avp(value)\r\n");
  sl_send_reply("123", "Custom Response");
  exit;
}

where P-Custom-Header appears in the 123 Custom Response reply.  The above
snippet would work only for a relayed request I believe, and since I
haven't received a network reply, I'm not sure how I could use an
onreply_route.  I'm stuck.



- Jeff
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From miha at softnet.si  Fri Jul  1 07:23:04 2016
From: miha at softnet.si (Miha)
Date: Fri, 1 Jul 2016 07:23:04 +0200
Subject: [OpenSIPS-Users] Issue with ACK and rtpproxy, setID
In-Reply-To: <5783BB89-736B-4AFA-9B12-9BD997DDBC50@inin.com>
References: <d6d01881-c228-a07f-2015-e26b1bb56b6c@softnet.si>
	<5783BB89-736B-4AFA-9B12-9BD997DDBC50@inin.com>
Message-ID: <46ec35b2-24c1-099a-355c-4037c3a64ed9@softnet.si>

Ben tnx.

make sense :)



br

miha

On 30/06/2016 15:17, Newlin, Ben wrote:
> AVPs are tied to a transaction, so the transaction must be matched before they will be available. You should use t_check_trans() to do this.
>
> However, I think this will not work for you because ACKs are their own transactions and I don?t believe they will have access to the AVPs from the INVITE transaction. If you need to store state information across multiple transactions, you will need to use the dialog module and the $dlg_val variables. These persist across the entire SIP call.
>
>
> Ben Newlin
>
> On 6/30/16, 8:06 AM, "users-bounces at lists.opensips.org on behalf of Miha" <users-bounces at lists.opensips.org on behalf of miha at softnet.si> wrote:
>
> HI
>
> I have two RTPproxies and doing spiral so that I can put them in chain.
>
> Before t_relay() I am setting avps (setID) so that I can do
> rtpproxy_answer latter if there is SDP in ACK. The issue is that avp is
> all the time null for ACK (for Initial invite avp was set).
>
> In TM module i set onreply_avp_mode to 1. Is there anything else I must
> do or do you suggest some other approche?
>
>
> tnx
>
> miha
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From alain.bieuzent at free.fr  Fri Jul  1 09:03:55 2016
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Fri, 01 Jul 2016 09:03:55 +0200
Subject: [OpenSIPS-Users] change_reply_status if failure route
In-Reply-To: <669771467304355@web18g.yandex.ru>
References: <669771467304355@web18g.yandex.ru>
Message-ID: <BA46924B-B34F-47AA-A4C4-F4419EFE275C@free.fr>

Hi, something like that ?

 

if ( t_check_status("487") ) {

????????????????????? send_reply("404","Not Found");

????????????????????? exit;

??????????????????????? }

 

De : <users-bounces at lists.opensips.org> au nom de ?????? ????? <achalkov at ya.ru>
R?pondre ? : OpenSIPS users mailling list <users at lists.opensips.org>
Date : jeudi 30 juin 2016 18:32
? : "users &gt;&gt; OpenSIPS users mailling list" <users at lists.opensips.org>
Objet : [OpenSIPS-Users] change_reply_status if failure route

 

hello everyone! 

Can someone explain me, could i use change_reply_status() if failure route or it hardcoded to use only in reply route or maybe help me with my problem: 

 

Case:

uac (1) <--> opensips (2) <--> uas (3)

 

step 1. 2 sends INVITE to 3

step 2. 3 sends 407 to 2

processing: Now 2 trying to assign avp's to use uac_auth() function, but in some case cant do it and 407's handling interrupts in main route by exit;

step 3. 2 sends ACK to 3

step 4. 2 sends 407 to 1

 


Could i somehow catch that 407 after step 2 and transform to something like 404 to send it instead of 407 to 1? How to do it?

 

_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users 

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From bogdan at opensips.org  Fri Jul  1 12:04:48 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 1 Jul 2016 13:04:48 +0300
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
Message-ID: <57764040.7050403@opensips.org>

Hello Alain,

You mean when comes to sent to ping a single destination, there are 
actually 2 OPTIONS sent out (on different interfaces) or it alternates 
(one ping is sent via eth0, the next one via eth1) ?

if you have a pcap to show the problem, please send it to me off-list.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.06.2016 12:49, Alain Bieuzent wrote:
>
> Hi all,
>
> I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.
>
> My server has 3 interfaces:
>
> eth0 with public IP
>
> eth0:0 with public IP provide via a keepalived script
>
> eth1 with a private IP
>
> Opensips listen on IP of eth0:0 and on eth1
>
> Mhomed=no
>
> Since i migrate to V2.1.3, load balancer probing is UP and down. After 
> checking traffic, i find for each gateway used by load balancer table, 
> there is 2 SIP OPTIONS sent.
>
> -First is sent with source IP of eth0:0
>
> -Second is sent with source IP of eth1
>
> Any idea ?
>
> Thanls
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Jul  1 12:51:40 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 1 Jul 2016 13:51:40 +0300
Subject: [OpenSIPS-Users] How to append header to stateless reply
In-Reply-To: <CACYJG3+U4czpLV6c+aiuLg39nw3QuDFpqSZbj4Td6YSpOd90TQ@mail.gmail.com>
References: <CACYJG3+U4czpLV6c+aiuLg39nw3QuDFpqSZbj4Td6YSpOd90TQ@mail.gmail.com>
Message-ID: <57764B3C.7080405@opensips.org>

Hi Jeff,

See append_to_reply() function:
http://www.opensips.org/html/docs/modules/2.2.x/sipmsgops.html#id249643

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 01.07.2016 05:15, Jeff Pyle wrote:
> Hello,
>
> How can I add a header to a stateless reply?  Or, if that's not 
> possible, to a transactional reply?
>
> I'm trying to do something like this:
>
> route {
>   ...
>   append_hf("P-Custom-Header: $avp(value)\r\n");
>   sl_send_reply("123", "Custom Response");
>   exit;
> }
>
> where P-Custom-Header appears in the 123 Custom Response reply.  The 
> above snippet would work only for a relayed request I believe, and 
> since I haven't received a network reply, I'm not sure how I could use 
> an onreply_route.  I'm stuck.
>
>
>
> - Jeff
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Jul  1 12:53:02 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 1 Jul 2016 13:53:02 +0300
Subject: [OpenSIPS-Users] change_reply_status if failure route
In-Reply-To: <669771467304355@web18g.yandex.ru>
References: <669771467304355@web18g.yandex.ru>
Message-ID: <57764B8E.2040800@opensips.org>

Hi Artem,

As Alain mention, just use t_reply() in failure route, to override the 
current reply.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 30.06.2016 19:32, ?????? ????? wrote:
> hello everyone!
> Can someone explain me, could i use change_reply_status() if failure 
> route or it hardcoded to use only in reply route or maybe help me with 
> my problem:
> Case:
> uac (1) <--> opensips (2) <--> uas (3)
> *step 1. 2 sends INVITE to 3*
> *step 2. 3 sends 407 to 2*
> processing: Now 2 trying to assign avp's to use uac_auth() function, 
> but in some case cant do it and 407's handling interrupts in main 
> route by exit;
> *step 3. 2 sends ACK to 3*
> *step 4. 2 sends 407 to 1*
> *
> *Could i somehow catch that 407 after step 2 and transform to 
> something like 404 to send it instead of 407 to 1? How to do it?
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From achalkov at ya.ru  Fri Jul  1 13:08:57 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Fri, 01 Jul 2016 14:08:57 +0300
Subject: [OpenSIPS-Users] change_reply_status if failure route
In-Reply-To: <57764B8E.2040800@opensips.org>
References: <669771467304355@web18g.yandex.ru> <57764B8E.2040800@opensips.org>
Message-ID: <3564171467371337@web7m.yandex.ru>

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From goup2010 at gmail.com  Fri Jul  1 13:31:42 2016
From: goup2010 at gmail.com (Dragomir Haralambiev)
Date: Fri, 1 Jul 2016 14:31:42 +0300
Subject: [OpenSIPS-Users] OpenSips and CLIR
Message-ID: <CADzurOLpmTmungiCMGfBSmVQLZskE=x_oa-eA5zE4_KekWhoVA@mail.gmail.com>

Hello,

How to activate CLIR from OpenSips?

Best regards,
Dragomir
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From tito at xsvoce.com  Fri Jul  1 15:23:10 2016
From: tito at xsvoce.com (Tito Cumpen)
Date: Fri, 1 Jul 2016 09:23:10 -0400
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <57726D11.5010904@opensips.org>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
Message-ID: <CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>

Bogdan,

Here is the backtrace:


Reading symbols from /usr/sbin/opensips...done.

[New LWP 2648]

[Thread debugging using libthread_db enabled]

Using host libthread_db library "/lib64/libthread_db.so.1".

Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips
-g opensips -M 1024 -f /etc'.

Program terminated with signal 11, Segmentation fault.

#0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
qm=0x7fadc0809010) at mem/f_malloc.c:187

187 *pf=n->u.nxt_free;

Missing separate debuginfos, use: debuginfo-install
cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 glibc-2.17-106.el7_2.4.x86_64
gmp-6.0.0-12.el7_1.x86_64 gnutls-3.3.8-14.el7_2.x86_64
keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.13.2-12.el7_2.x86_64
libcom_err-1.42.9-7.el7.x86_64 libcurl-7.29.0-25.el7.centos.x86_64
libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64
libgcrypt-1.5.3-12.el7_1.1.x86_64 libgpg-error-1.12-3.el7.x86_64
libidn-1.28-4.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64
librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64
libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64
libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64
nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64
nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64

(gdb) bt full

#0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
qm=0x7fadc0809010) at mem/f_malloc.c:187

        pf = 0x0

        hash = 2051

#1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
mem/f_malloc.c:415

        frag = 0x7fadc0898210

        n = <optimized out>

        hash = 2051

        __FUNCTION__ = "fm_malloc"

#2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
../proto_ws/ws_common.h:576

        size = <optimized out>

        req = <optimized out>

        ret_code = WS_ERR_NONE

        bk = <optimized out>

        msg_len = <optimized out>

        local_rcv = {src_ip = {af = 3082146304, len = 32685, u = {addrl =
{34359801655, 140720703251544}, addr32 = {63287, 8, 394765400, 32764},
addr16 = {63287, 0, 8, 0, 42072, 6023, 32764, 0},

              addr =
"7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}}, dst_ip = {af =
3080002780, len = 32685, u = {addrl = {4, 63119970000}, addr32 = {4, 0,
2990427856, 14}, addr16 = {4, 0, 0,

                0, 20176, 45630, 14, 0}, addr =
"\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}}, src_port =
52976, dst_port = 49287, proto = 32685, proto_reserved1 = -1214962633,

          proto_reserved2 = 32685, src_su = {s = {sa_family = 2, sa_data =
")\350n\343\276C\360#%\275\016\000\000"}, sin = {sin_family = 2, sin_port =
59433, sin_addr = {s_addr = 1136583534},

              sin_zero = "\360#%\275\016\000\000"}, sin6 = {sin6_family =
2, sin6_port = 59433, sin6_flowinfo = 1136583534, sin6_addr = {__in6_u = {

                  __u6_addr8 =
"\360#%\275\016\000\000\000\360?\300\255\177\000", __u6_addr16 = {9200,
48421, 14, 0, 52976, 49287, 32685, 0}, __u6_addr32 = {3173327856, 14,
3230125808, 32685}}},

              sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}

        newreq = <optimized out>

        msg_buf = <optimized out>

#3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized out>) at
proto_wss.c:428

        size = <optimized out>

        __FUNCTION__ = "wss_read_req"

#4  0x000000000059c374 in handle_io (fm=<optimized out>, idx=idx at entry=0,
event_type=event_type at entry=1) at net/net_tcp_proc.c:205

        ret = 0

        n = <optimized out>

        con = 0x7fadbeb3f718

        s = 57

        rw = <optimized out>

        resp = <optimized out>

        response = {140384203757736, 1}

        __FUNCTION__ = "handle_io"

#5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized out>,
t=<optimized out>, repeat=<optimized out>) at net/../io_wait_loop.h:221

        ret = <optimized out>

        e = <optimized out>

        n = 1

        r = 0

#6  tcp_worker_proc (unix_sock=<optimized out>) at net/net_tcp_proc.c:312

        __FUNCTION__ = "tcp_worker_proc"

#7  0x00000000005a7fc9 in tcp_start_processes (chd_rank=chd_rank at entry=0x8417e0
<chd_rank.11018>, startup_done=startup_done at entry=0x7fadbe978738) at
net/net_tcp.c:1761

        r = 5

        reader_fd = {50, 52}

        pid = 0

        load_p = 0x7fadbe979398

        __FUNCTION__ = "tcp_start_processes"

#8  0x0000000000419ef5 in main_loop () at main.c:677

        startup_done = 0x7fadbe978738

        chd_rank = 18

#9  main (argc=<optimized out>, argv=<optimized out>) at main.c:1249

        cfg_stream = <optimized out>

        c = <optimized out>

        r = <optimized out>

        tmp = 0x7ffc1787bf65 ""

        tmp_len = <optimized out>

        port = <optimized out>

        proto = <optimized out>

        protos_no = <optimized out>

        options = 0x5dae60 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"

---Type <return> to continue, or q <return> to quit---

        ret = -1

        seed = 995413301

        rfd = <optimized out>

        __FUNCTION__ = "main"



currently using version: opensips 2.3.0-dev (x86_64/linux)

flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT

ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.

git revision: 92434ef

main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5

On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Tito,
>
> If opensips crashes, were you able to extract a backtrace from the core
> file(s) ?
>
> Thanks and regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 27.06.2016 21:20, Tito Cumpen wrote:
>
> Group,
>
>
> I am experiencing strange behavior when configuring sip tls on port 443.
> At time opensips crashes or stops accepting new connections. Here are the
> tcp configs I am using:
>
> #disable_tcp=no
>
> tcp_connection_lifetime=3600
>
> tcp_connect_timeout=3
>
> tcp_keepidle = 30
>
> tcp_keepinterval = 5
>
> tcp_keepalive = 1
>
> tcp_keepcount = 5
>
> tcp_max_msg_time = 8
> tcp_children=10
>
>
> Any idea what would case this? I am assuming there are probes out in the
> internet that eventually make opensips crash?
>
> Thanks,
> Tito
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From jim at devito.cc  Fri Jul  1 15:31:35 2016
From: jim at devito.cc (Jim DeVito)
Date: Fri, 01 Jul 2016 06:31:35 -0700
Subject: [OpenSIPS-Users] OpenSips and CLIR
In-Reply-To: <CADzurOLpmTmungiCMGfBSmVQLZskE=x_oa-eA5zE4_KekWhoVA@mail.gmail.com>
References: <CADzurOLpmTmungiCMGfBSmVQLZskE=x_oa-eA5zE4_KekWhoVA@mail.gmail.com>
Message-ID: <f7de5e48ad054b2a8513227d58ea6f60@mail.devito.cc>

Depends how your upstream expects you to handle privacy. In my case it 
is enough to set the user part of the from uri to "anonymous" and use 
insert_hf("Privacy: id\r\n", "Allow-Events"); to add the privacy header.

---
Jim DeVito

On 2016-07-01 04:31, Dragomir Haralambiev wrote:
> Hello,
> 
> How to activate CLIR from OpenSips?
> 
> Best regards,
> Dragomir
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From tito at xsvoce.com  Fri Jul  1 17:44:10 2016
From: tito at xsvoce.com (Tito Cumpen)
Date: Fri, 1 Jul 2016 11:44:10 -0400
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
Message-ID: <CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>

Bogdan,


Correction I was using :

version: opensips 2.2.0 (x86_64/linux)

flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT

ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.

git revision: d835721

main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5





On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen <tito at xsvoce.com> wrote:

> Bogdan,
>
> Here is the backtrace:
>
>
> Reading symbols from /usr/sbin/opensips...done.
>
> [New LWP 2648]
>
> [Thread debugging using libthread_db enabled]
>
> Using host libthread_db library "/lib64/libthread_db.so.1".
>
> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips
> -g opensips -M 1024 -f /etc'.
>
> Program terminated with signal 11, Segmentation fault.
>
> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
> qm=0x7fadc0809010) at mem/f_malloc.c:187
>
> 187 *pf=n->u.nxt_free;
>
> Missing separate debuginfos, use: debuginfo-install
> cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 glibc-2.17-106.el7_2.4.x86_64
> gmp-6.0.0-12.el7_1.x86_64 gnutls-3.3.8-14.el7_2.x86_64
> keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.13.2-12.el7_2.x86_64
> libcom_err-1.42.9-7.el7.x86_64 libcurl-7.29.0-25.el7.centos.x86_64
> libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64
> libgcrypt-1.5.3-12.el7_1.1.x86_64 libgpg-error-1.12-3.el7.x86_64
> libidn-1.28-4.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64
> librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64
> libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64
> libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64
> nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64
> nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
> nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
> openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
> pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
> xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64
>
> (gdb) bt full
>
> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
> qm=0x7fadc0809010) at mem/f_malloc.c:187
>
>         pf = 0x0
>
>         hash = 2051
>
> #1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
> mem/f_malloc.c:415
>
>         frag = 0x7fadc0898210
>
>         n = <optimized out>
>
>         hash = 2051
>
>         __FUNCTION__ = "fm_malloc"
>
> #2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
> ../proto_ws/ws_common.h:576
>
>         size = <optimized out>
>
>         req = <optimized out>
>
>         ret_code = WS_ERR_NONE
>
>         bk = <optimized out>
>
>         msg_len = <optimized out>
>
>         local_rcv = {src_ip = {af = 3082146304, len = 32685, u = {addrl =
> {34359801655, 140720703251544}, addr32 = {63287, 8, 394765400, 32764},
> addr16 = {63287, 0, 8, 0, 42072, 6023, 32764, 0},
>
>               addr =
> "7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}}, dst_ip = {af =
> 3080002780, len = 32685, u = {addrl = {4, 63119970000}, addr32 = {4, 0,
> 2990427856, 14}, addr16 = {4, 0, 0,
>
>                 0, 20176, 45630, 14, 0}, addr =
> "\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}}, src_port =
> 52976, dst_port = 49287, proto = 32685, proto_reserved1 = -1214962633,
>
>           proto_reserved2 = 32685, src_su = {s = {sa_family = 2, sa_data =
> ")\350n\343\276C\360#%\275\016\000\000"}, sin = {sin_family = 2, sin_port =
> 59433, sin_addr = {s_addr = 1136583534},
>
>               sin_zero = "\360#%\275\016\000\000"}, sin6 = {sin6_family =
> 2, sin6_port = 59433, sin6_flowinfo = 1136583534, sin6_addr = {__in6_u = {
>
>                   __u6_addr8 =
> "\360#%\275\016\000\000\000\360?\300\255\177\000", __u6_addr16 = {9200,
> 48421, 14, 0, 52976, 49287, 32685, 0}, __u6_addr32 = {3173327856, 14,
> 3230125808, 32685}}},
>
>               sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}
>
>         newreq = <optimized out>
>
>         msg_buf = <optimized out>
>
> #3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized out>) at
> proto_wss.c:428
>
>         size = <optimized out>
>
>         __FUNCTION__ = "wss_read_req"
>
> #4  0x000000000059c374 in handle_io (fm=<optimized out>, idx=idx at entry=0,
> event_type=event_type at entry=1) at net/net_tcp_proc.c:205
>
>         ret = 0
>
>         n = <optimized out>
>
>         con = 0x7fadbeb3f718
>
>         s = 57
>
>         rw = <optimized out>
>
>         resp = <optimized out>
>
>         response = {140384203757736, 1}
>
>         __FUNCTION__ = "handle_io"
>
> #5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized out>,
> t=<optimized out>, repeat=<optimized out>) at net/../io_wait_loop.h:221
>
>         ret = <optimized out>
>
>         e = <optimized out>
>
>         n = 1
>
>         r = 0
>
> #6  tcp_worker_proc (unix_sock=<optimized out>) at net/net_tcp_proc.c:312
>
>         __FUNCTION__ = "tcp_worker_proc"
>
> #7  0x00000000005a7fc9 in tcp_start_processes (chd_rank=chd_rank at entry=0x8417e0
> <chd_rank.11018>, startup_done=startup_done at entry=0x7fadbe978738) at
> net/net_tcp.c:1761
>
>         r = 5
>
>         reader_fd = {50, 52}
>
>         pid = 0
>
>         load_p = 0x7fadbe979398
>
>         __FUNCTION__ = "tcp_start_processes"
>
> #8  0x0000000000419ef5 in main_loop () at main.c:677
>
>         startup_done = 0x7fadbe978738
>
>         chd_rank = 18
>
> #9  main (argc=<optimized out>, argv=<optimized out>) at main.c:1249
>
>         cfg_stream = <optimized out>
>
>         c = <optimized out>
>
>         r = <optimized out>
>
>         tmp = 0x7ffc1787bf65 ""
>
>         tmp_len = <optimized out>
>
>         port = <optimized out>
>
>         proto = <optimized out>
>
>         protos_no = <optimized out>
>
>         options = 0x5dae60 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"
>
> ---Type <return> to continue, or q <return> to quit---
>
>         ret = -1
>
>         seed = 995413301
>
>         rfd = <optimized out>
>
>         __FUNCTION__ = "main"
>
>
>
> currently using version: opensips 2.3.0-dev (x86_64/linux)
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>
> git revision: 92434ef
>
> main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>
> On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Tito,
>>
>> If opensips crashes, were you able to extract a backtrace from the core
>> file(s) ?
>>
>> Thanks and regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 27.06.2016 21:20, Tito Cumpen wrote:
>>
>> Group,
>>
>>
>> I am experiencing strange behavior when configuring sip tls on port 443.
>> At time opensips crashes or stops accepting new connections. Here are the
>> tcp configs I am using:
>>
>> #disable_tcp=no
>>
>> tcp_connection_lifetime=3600
>>
>> tcp_connect_timeout=3
>>
>> tcp_keepidle = 30
>>
>> tcp_keepinterval = 5
>>
>> tcp_keepalive = 1
>>
>> tcp_keepcount = 5
>>
>> tcp_max_msg_time = 8
>> tcp_children=10
>>
>>
>> Any idea what would case this? I am assuming there are probes out in the
>> internet that eventually make opensips crash?
>>
>> Thanks,
>> Tito
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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From millennium.bug at gmail.com  Fri Jul  1 19:11:14 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Fri, 1 Jul 2016 22:11:14 +0500
Subject: [OpenSIPS-Users] opensips mysql database connectivity
Message-ID: <CAArAGKnkCoW=at=T_x0PV0OXNndAHaL2dOGr-OJ2fS=8kw8nDQ@mail.gmail.com>

Hello all,

I
? am using db_mysql module on opensips version 2.1.3

Is there a way to limit the number of db connections that opensips
initializes with?
Also, how can we make non-persistent connections mysql connections with
opensips??


Regards,
Owais
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From nabeelshikder at gmail.com  Sat Jul  2 00:48:27 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Fri, 1 Jul 2016 23:48:27 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <5774E3ED.6020301@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
Message-ID: <CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>

Hi,

Adding 'limit 1' or 'limit 5' to the supusers mysql view resolves part of
the error, but I don't understand why that is and whether this is correct
for the setup. Maybe something to do with connection pooling?

Now the following errors remain:

[Jun 30 01:07:53] NOTICE[17067][C-00000000] chan_sip.c: Call from '
> +447867958678' (162.248.6.216:12351 <http://162.249.6.206:12221/>) to
> extension 'VMR_+447479189410' rejected because extension not found in
> context 'default'.
> [Jun 30 01:07:53] WARNING[17112] res_config_odbc.c: SQL Prepare
> failed![SELECT * FROM sipusers WHERE name = ? AND host = ?]
> [Jun 30 01:07:53] WARNING[17112] res_odbc.c: Connection is down attempting
> to reconnect.


I am using OpenSIPS 2.2, not 2.3 as stated earlier.

Nabeel

On 30 June 2016 at 10:18, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:

> Hi Nabeel,
>
> The "sipusers" mysql view (as per
> http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8#toc7
> ) has both the name and host fields - not sure why that query may fail.....
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 30.06.2016 07:19, Nabeel wrote:
>
> Hi Bogdan,
>
> I was able to install the latest versions of Asterisk (13.1) and Opensips
> (2.3) according to the tutorial, but when attempting to leave a voicemail I
> get the following errors:
>
>
>> [Jun 30 01:07:53] NOTICE[17067][C-00000000] chan_sip.c: Call from '
>> +447867958678' (162.249.6.206:12221) to extension 'VMR_+447479189410'
>> rejected because extension not found in context 'default'.
>> [Jun 30 01:07:53] WARNING[17112] res_odbc.c: SetConnectAttr (Txn
>> isolation) returned an error: HY000: [MySQL][ODBC 5.2(w) Driver]You have an
>> error in your SQL syntax; check the manual that corresponds to your MariaDB
>> server version for the right syntax to use near '7' at line 1
>> [Jun 30 01:07:53] WARNING[17112] res_config_odbc.c: SQL Prepare
>> failed![SELECT * FROM sipusers WHERE name = ? AND host = ?]
>> [Jun 30 01:07:53] WARNING[17112] res_odbc.c: Connection is down
>> attempting to reconnect...
>>
>
>
> Also I had to change 'nat=yes' to 'nat=force_rport,comedia' as it is
> deprecated.
>
> Nabeel
>
>
> On 14 June 2016 at 11:08, Bogdan-Andrei Iancu < <bogdan at opensips.org>
> bogdan at opensips.org> wrote:
>
>> Hi Nabeel,
>>
>> We will update the tutorial for 2.2, but it should still match. Give it a
>> try and if you hit issues, just let me know.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 12.06.2016 10:18, Nabeel wrote:
>>
>> Hi,
>>
>> I will be following this tutorial to integrate OpenSIPS and Asterisk:
>>
>>
>> http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8
>>
>> The tutorial mentions the use of OpenSIPS version 1.8 and Asterisk
>> version 1.8. I would like to know if I can use the latest versions of
>> OpenSIPS and Asterisk instead? Have there been changes to database
>> structure which can cause problems?
>>
>> Nabeel
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>
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From nabeelshikder at gmail.com  Sat Jul  2 03:23:57 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sat, 2 Jul 2016 02:23:57 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
Message-ID: <CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>

The tutorial contains a mistake where the priority ordering in
extensions.conf should start with 1, not n:

; Voicemail
> exten => _VMR_.,1,Ringing
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Answer
> exten => _VMR_.,n,Wait(1)
> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
> exten => _VMR_.,n,Hangup

; Allow users to call their Voicemail directly
> exten => VM_pickup,1,Ringing
> exten => VM_pickup,n,wait(1)
> exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
> exten => VM_pickup,n,Hangup
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From nabeelshikder at gmail.com  Sat Jul  2 03:36:13 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sat, 2 Jul 2016 02:36:13 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
Message-ID: <CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>

The issue in my last Email has solved the error about missing extension.
Now the following errors remain:

[Jul  2 02:29:18] WARNING[18226][C-00000001]: res_config_odbc.c:117
> custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ?
> AND callbackextension = ? AND port = ?]
> [Jul  2 02:29:22] WARNING[18269][C-00000000]: app.c:1633
> __ast_play_and_record: No audio available on SIP/domain.com-00000000??



On 2 July 2016 at 02:23, Nabeel <nabeelshikder at gmail.com> wrote:

> The tutorial contains a mistake where the priority ordering in
> extensions.conf should start with 1, not n:
>
> ; Voicemail
>> exten => _VMR_.,1,Ringing
>> exten => _VMR_.,n,Wait(1)
>> exten => _VMR_.,n,Answer
>> exten => _VMR_.,n,Wait(1)
>> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>> exten => _VMR_.,n,Hangup
>
> ; Allow users to call their Voicemail directly
>> exten => VM_pickup,1,Ringing
>> exten => VM_pickup,n,wait(1)
>> exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>> exten => VM_pickup,n,Hangup
>
>
>
>
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From nabeelshikder at gmail.com  Sat Jul  2 04:29:59 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sat, 2 Jul 2016 03:29:59 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
Message-ID: <CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>

In the last error message,* '**callbackextension = ?' *suggested that this
column is missing from the sipusers mysql view. So I added this column to
the view and now that error has been resolved. Only the following error
remains now:

[Jul  2 03:25:48] WARNING[19330][C-00000005]: app.c:1633
> __ast_play_and_record: No audio available on SIP/domain.com-00000005??






On 2 July 2016 at 02:36, Nabeel <nabeelshikder at gmail.com> wrote:

> The issue in my last Email has solved the error about missing extension.
> Now the following errors remain:
>
> [Jul  2 02:29:18] WARNING[18226][C-00000001]: res_config_odbc.c:117
>> custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ?
>> AND callbackextension = ? AND port = ?]
>> [Jul  2 02:29:22] WARNING[18269][C-00000000]: app.c:1633
>> __ast_play_and_record: No audio available on SIP/domain.com-00000000??
>
>
>
> On 2 July 2016 at 02:23, Nabeel <nabeelshikder at gmail.com> wrote:
>
>> The tutorial contains a mistake where the priority ordering in
>> extensions.conf should start with 1, not n:
>>
>> ; Voicemail
>>> exten => _VMR_.,1,Ringing
>>> exten => _VMR_.,n,Wait(1)
>>> exten => _VMR_.,n,Answer
>>> exten => _VMR_.,n,Wait(1)
>>> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>>> exten => _VMR_.,n,Hangup
>>
>> ; Allow users to call their Voicemail directly
>>> exten => VM_pickup,1,Ringing
>>> exten => VM_pickup,n,wait(1)
>>> exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>>> exten => VM_pickup,n,Hangup
>>
>>
>>
>>
>
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From nabeelshikder at gmail.com  Sat Jul  2 14:41:00 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sat, 2 Jul 2016 13:41:00 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
Message-ID: <CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>

In the latest version of Asterisk, there is a new file voicemail.conf which
must be configured correctly for voicemail, but the tutorial does not
mention this file at all. Please let me know how to configure this file for
integration with OpenSIPS.

Nabeel
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From daniel.zanutti at gmail.com  Sat Jul  2 23:57:10 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Sat, 2 Jul 2016 18:57:10 -0300
Subject: [OpenSIPS-Users] ACC failover solution
Message-ID: <CAN0jgiPmAJk0sz3ex10FKj4XPoHhDjrU9u433BNNS6Y8v7J2xQ@mail.gmail.com>

Hi

I need some help to provide a failover solution to do accounting on our
calls.

If ACC fails to write on MySQL accounting system (Like MySQL), the record
is lost. Is there any failover solution on the ACC module, to record on
another kind of DB, only if MySQL failed?

I know I could write on both but it's a waste of resources, I wanted to
record on another kind of DB only if the first (or primary) failed.

Thanks in advance!
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From jarrod at unixc.org  Sun Jul  3 00:07:42 2016
From: jarrod at unixc.org (Jarrod Baumann)
Date: Sat, 2 Jul 2016 17:07:42 -0500
Subject: [OpenSIPS-Users] ACC failover solution
In-Reply-To: <CAN0jgiPmAJk0sz3ex10FKj4XPoHhDjrU9u433BNNS6Y8v7J2xQ@mail.gmail.com>
References: <CAN0jgiPmAJk0sz3ex10FKj4XPoHhDjrU9u433BNNS6Y8v7J2xQ@mail.gmail.com>
Message-ID: <2477338F-A5B7-4BD0-B1EF-6A55894C7ABC@unixc.org>

Daniel,

You want to look at db_virtual (http://www.opensips.org/html/docs/modules/2.2.x/db_virtual.html <http://www.opensips.org/html/docs/modules/2.2.x/db_virtual.html>)

You can define failover database as such:
modparam("db_virtual", "db_urls", "define accounting FAILOVER")
modparam("db_virtual", "db_urls", "mysql://opensips:opensipsrw at host1/testa")
modparam("db_virtual", "db_urls", "mysql://opensips:opensipsrw at host2/testa")

Your acc db url would look something like this:
modparam("acc", "db_url", "virtual://accounting?)

jarrod

> On Jul 2, 2016, at 4:57 PM, Daniel Zanutti <d
> aniel.zanutti at gmail.com> wrote:
> 
> Hi
> 
> I need some help to provide a failover solution to do accounting on our calls.
> 
> If ACC fails to write on MySQL accounting system (Like MySQL), the record is lost. Is there any failover solution on the ACC module, to record on another kind of DB, only if MySQL failed?
> 
> I know I could write on both but it's a waste of resources, I wanted to record on another kind of DB only if the first (or primary) failed.
> 
> Thanks in advance!
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From daniel.zanutti at gmail.com  Sun Jul  3 00:28:54 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Sat, 2 Jul 2016 19:28:54 -0300
Subject: [OpenSIPS-Users] ACC failover solution
In-Reply-To: <2477338F-A5B7-4BD0-B1EF-6A55894C7ABC@unixc.org>
References: <CAN0jgiPmAJk0sz3ex10FKj4XPoHhDjrU9u433BNNS6Y8v7J2xQ@mail.gmail.com>
	<2477338F-A5B7-4BD0-B1EF-6A55894C7ABC@unixc.org>
Message-ID: <CAN0jgiMbLFBr4JRj5LXoAvDoYLyevEGVK2dWdz8MRp67ZCdPgg@mail.gmail.com>

Great! I'll take a look

On Sat, Jul 2, 2016 at 7:07 PM, Jarrod Baumann <jarrod at unixc.org> wrote:

> Daniel,
>
> You want to look at db_virtual (
> http://www.opensips.org/html/docs/modules/2.2.x/db_virtual.html)
>
> You can define failover database as such:
> modparam("db_virtual", "db_urls", "define accounting FAILOVER")
> modparam("db_virtual", "db_urls", "mysql://opensips:opensipsrw at host1/testa
> ")
> modparam("db_virtual", "db_urls", "mysql://opensips:opensipsrw at host2/testa
> ")
>
> Your acc db url would look something like this:
> modparam("acc", "db_url", "virtual://accounting?)
>
> jarrod
>
> On Jul 2, 2016, at 4:57 PM, Daniel Zanutti <d
>
> aniel.zanutti at gmail.com> wrote:
>
> Hi
>
> I need some help to provide a failover solution to do accounting on our
> calls.
>
> If ACC fails to write on MySQL accounting system (Like MySQL), the record
> is lost. Is there any failover solution on the ACC module, to record on
> another kind of DB, only if MySQL failed?
>
> I know I could write on both but it's a waste of resources, I wanted to
> record on another kind of DB only if the first (or primary) failed.
>
> Thanks in advance!
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From nabeelshikder at gmail.com  Sun Jul  3 21:59:03 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sun, 3 Jul 2016 20:59:03 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
	<CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
Message-ID: <CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>

The last error message has been solved by removing the following lines from
opensips.cfg:

     if (!db_does_uri_exist()) {
>                        send_reply("420","Bad Extension");
>                        exit;
>                }
>


>                t_newtran();
>                t_reply("480", "Temporarily Unavailable");



Now voicemail seems to be working, but only if manually adding users to the
voicemail.conf file. So the following questions remain:

1. Adding "limit = 1" to sipusers mysql view resolves a database error. Is
this the correct way to fix the error for voicemail integration?

2. How can voicemail.conf file be configured to use variable substitution
for all users in the OpenSIPS subscriber table?

Nabeel

On 2 July 2016 at 13:41, Nabeel <nabeelshikder at gmail.com> wrote:

> In the latest version of Asterisk, there is a new file voicemail.conf
> which must be configured correctly for voicemail, but the tutorial does not
> mention this file at all. Please let me know how to configure this file for
> integration with OpenSIPS.
>
> Nabeel
>
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From govoiper at gmail.com  Sun Jul  3 22:20:52 2016
From: govoiper at gmail.com (SamyGo)
Date: Sun, 3 Jul 2016 16:20:52 -0400
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
	<CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
	<CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>
Message-ID: <CAJUJwtgP=+VxEiUBcDWdLpm4abSH_eHGk=y1GZD_VYfOuqM90Q@mail.gmail.com>

Hi Nabeel,

Point 1 I cant imagine how those lines possibly relate to no media error in
asterisk, I guess it depends on your config setup.

The logical answer to your point 2 would be Asterisk realtime. However this
is not going to be as staraight forward as making asterisk use subscriber
table for voicemail since thats not how asterisk realtime works.

You can tell asterisk which table to point to when finding a vmail box but
that table needs to have voicemail related columns.
There could be more complicated ways to instruct asterisk to do what you
want can be, for example, ARI /AGI/custom dialplan and IVR. However, all of
those should be discussed in Asterisk mailing list.

The OpenSIPS integration guide did serve it's purpose in showing you the
path, agreed that it should be updated but I believe OpenSIPS team is doing
much bigger things. Your efforts here might be very helpful to alot of
people.

Regards,
Sammy
The last error message has been solved by removing the following lines from
opensips.cfg:

     if (!db_does_uri_exist()) {
>                        send_reply("420","Bad Extension");
>                        exit;
>                }
>


>                t_newtran();
>                t_reply("480", "Temporarily Unavailable");



Now voicemail seems to be working, but only if manually adding users to the
voicemail.conf file. So the following questions remain:

1. Adding "limit = 1" to sipusers mysql view resolves a database error. Is
this the correct way to fix the error for voicemail integration?

2. How can voicemail.conf file be configured to use variable substitution
for all users in the OpenSIPS subscriber table?

Nabeel

On 2 July 2016 at 13:41, Nabeel <nabeelshikder at gmail.com> wrote:

> In the latest version of Asterisk, there is a new file voicemail.conf
> which must be configured correctly for voicemail, but the tutorial does not
> mention this file at all. Please let me know how to configure this file for
> integration with OpenSIPS.
>
> Nabeel
>


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From nabeelshikder at gmail.com  Sun Jul  3 22:51:20 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Sun, 3 Jul 2016 21:51:20 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CAJUJwtgP=+VxEiUBcDWdLpm4abSH_eHGk=y1GZD_VYfOuqM90Q@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
	<CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
	<CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>
	<CAJUJwtgP=+VxEiUBcDWdLpm4abSH_eHGk=y1GZD_VYfOuqM90Q@mail.gmail.com>
Message-ID: <CA+vx6KKx34SjzmgSE=9L0xwosxDD41GTecL5aU8-9t2Gz5w4Xg@mail.gmail.com>

Hi Samy,

Point 1 I cant imagine how those lines possibly relate to no media error in
> asterisk, I guess it depends on your config setup.


In point 1 I was referring to this error:

WARNING[17112] res_odbc.c: SetConnectAttr (Txn isolation) returned an
> error: HY000: [MySQL][ODBC 5.2(w) Driver]You have an error in your SQL
> syntax; check the manual that corresponds to your MariaDB server version
> for the right syntax to use near '7' at line 1


The above error was solved by adding "limit=1" to the sipusers mysql view.
Regarding the lines you are referring to -- removing them from the config
solved the 'no media error' because those reply messages were interfering
with the voicemail reply.

About your suggestion to use 'Asterisk Realtime', I thought that is
precisely what the tutorial is describing - how to integrate Asterisk
Realtime:

"This tutorial presents the concept and implementation of a realtime
integration of OpenSIPS SIP server and Asterisk media server."

I am a bit puzzled by your suggestion, but I will try asking in the
Asterisk mailing list.

Nabeel
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From millennium.bug at gmail.com  Mon Jul  4 07:03:35 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Mon, 4 Jul 2016 10:03:35 +0500
Subject: [OpenSIPS-Users] opensips mysql database connectivity
In-Reply-To: <CAArAGKnkCoW=at=T_x0PV0OXNndAHaL2dOGr-OJ2fS=8kw8nDQ@mail.gmail.com>
References: <CAArAGKnkCoW=at=T_x0PV0OXNndAHaL2dOGr-OJ2fS=8kw8nDQ@mail.gmail.com>
Message-ID: <CAArAGKkvYvU=t7TSphJxRUCNj2k1U-w9kVrdafFswqRe_4R7Lw@mail.gmail.com>

Does OpenSIPS offer DB connection pooling?

On Fri, Jul 1, 2016 at 10:11 PM, Owais Ahmad <millennium.bug at gmail.com>
wrote:

> Hello all,
>
> I
> ? am using db_mysql module on opensips version 2.1.3
>
> Is there a way to limit the number of db connections that opensips
> initializes with?
> Also, how can we make non-persistent connections mysql connections with
> opensips??
>
>
> Regards,
> Owais
>
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From bogdan at opensips.org  Mon Jul  4 09:46:44 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 10:46:44 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
Message-ID: <577A1464.4080100@opensips.org>

Hi,

This kind of ordering is valid in older versions of Asterisk. Maybe not 
anymore in the newer versions.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02.07.2016 04:23, Nabeel wrote:
> The tutorial contains a mistake where the priority ordering in 
> extensions.conf should start with 1, not n:
>
>     ; Voicemail
>     exten => _VMR_.,1,Ringing
>     exten => _VMR_.,n,Wait(1)
>     exten => _VMR_.,n,Answer
>     exten => _VMR_.,n,Wait(1)
>     exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>     exten => _VMR_.,n,Hangup 
>
>     ; Allow users to call their Voicemail directly
>     exten => VM_pickup,1,Ringing
>     exten => VM_pickup,n,wait(1)
>     exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>     exten => VM_pickup,n,Hangup
>
>
>

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From bogdan at opensips.org  Mon Jul  4 09:48:15 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 10:48:15 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
Message-ID: <577A14BF.6060602@opensips.org>

Hi,

What is the definition you used for this new column ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02.07.2016 05:29, Nabeel wrote:
> In the last error message,/'//callbackextension = ?' /suggested that 
> this column is missing from the sipusers mysql view. So I added this 
> column to the view and now that error has been resolved. Only the 
> following error remains now:
>
>     [Jul  2 03:25:48] WARNING[19330][C-00000005]: app.c:1633
>     __ast_play_and_record: No audio available on SIP/domain.com-00000005??
>
>
>
>
>
>
> On 2 July 2016 at 02:36, Nabeel <nabeelshikder at gmail.com 
> <mailto:nabeelshikder at gmail.com>> wrote:
>
>     The issue in my last Email has solved the error about missing
>     extension. Now the following errors remain:
>
>         [Jul  2 02:29:18] WARNING[18226][C-00000001]:
>         res_config_odbc.c:117 custom_prepare: SQL Prepare
>         failed![SELECT * FROM sipusers WHERE host = ? AND
>         callbackextension = ? AND port = ?]
>         [Jul  2 02:29:22] WARNING[18269][C-00000000]: app.c:1633
>         __ast_play_and_record: No audio available on
>         SIP/domain.com-00000000??
>
>
>
>     On 2 July 2016 at 02:23, Nabeel <nabeelshikder at gmail.com
>     <mailto:nabeelshikder at gmail.com>> wrote:
>
>         The tutorial contains a mistake where the priority ordering in
>         extensions.conf should start with 1, not n:
>
>             ; Voicemail
>             exten => _VMR_.,1,Ringing
>             exten => _VMR_.,n,Wait(1)
>             exten => _VMR_.,n,Answer
>             exten => _VMR_.,n,Wait(1)
>             exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>             exten => _VMR_.,n,Hangup 
>
>             ; Allow users to call their Voicemail directly
>             exten => VM_pickup,1,Ringing
>             exten => VM_pickup,n,wait(1)
>             exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>             exten => VM_pickup,n,Hangup
>
>
>
>
>

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From bogdan at opensips.org  Mon Jul  4 09:51:04 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 10:51:04 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
	<CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
Message-ID: <577A1568.8080504@opensips.org>

Hi,

the voicemail.conf file exists in almost all asterisk versions. But if 
you use the odbc storage for voicemail, you do not need this file at all.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 02.07.2016 15:41, Nabeel wrote:
>
> In the latest version of Asterisk, there is a new file voicemail.conf 
> which must be configured correctly for voicemail, but the tutorial 
> does not mention this file at all. Please let me know how to configure 
> this file for integration with OpenSIPS.
>
> Nabeel
>



From nabeelshikder at gmail.com  Mon Jul  4 10:38:43 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Mon, 4 Jul 2016 09:38:43 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <577A14BF.6060602@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
Message-ID: <CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>

Hi Bogdan,

I just added the column to the view by adding "NULL AS `callbackextension`"
to the SQL view definition. I haven't linked the column to the subscriber
column, so this may not be the correct definition. However, it got rid of
the error.

About the voicemail.conf file, when I attempted to leave a voicemail to a
number in subscriber table, I got an error telling me that the user/mailbox
does not exist in voicemail.conf file. After adding that number manually to
the file, voicemail was working for that number. I think this may also mean
that the integration is not working correctly in my current setup.

Nabeel
On 4 Jul 2016 8:48 a.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org> wrote:

> Hi,
>
> What is the definition you used for this new column ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02.07.2016 05:29, Nabeel wrote:
>
> In the last error message,* '**callbackextension = ?' *suggested that
> this column is missing from the sipusers mysql view. So I added this column
> to the view and now that error has been resolved. Only the following error
> remains now:
>
> [Jul  2 03:25:48] WARNING[19330][C-00000005]: app.c:1633
>> __ast_play_and_record: No audio available on SIP/domain.com-00000005??
>
>
>
>
>
>
> On 2 July 2016 at 02:36, Nabeel <nabeelshikder at gmail.com> wrote:
>
>> The issue in my last Email has solved the error about missing extension.
>> Now the following errors remain:
>>
>> [Jul  2 02:29:18] WARNING[18226][C-00000001]: res_config_odbc.c:117
>>> custom_prepare: SQL Prepare failed![SELECT * FROM sipusers WHERE host = ?
>>> AND callbackextension = ? AND port = ?]
>>> [Jul  2 02:29:22] WARNING[18269][C-00000000]: app.c:1633
>>> __ast_play_and_record: No audio available on SIP/domain.com-00000000??
>>
>>
>>
>> On 2 July 2016 at 02:23, Nabeel <nabeelshikder at gmail.com> wrote:
>>
>>> The tutorial contains a mistake where the priority ordering in
>>> extensions.conf should start with 1, not n:
>>>
>>> ; Voicemail
>>>> exten => _VMR_.,1,Ringing
>>>> exten => _VMR_.,n,Wait(1)
>>>> exten => _VMR_.,n,Answer
>>>> exten => _VMR_.,n,Wait(1)
>>>> exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>>>> exten => _VMR_.,n,Hangup
>>>
>>> ; Allow users to call their Voicemail directly
>>>> exten => VM_pickup,1,Ringing
>>>> exten => VM_pickup,n,wait(1)
>>>> exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>>>> exten => VM_pickup,n,Hangup
>>>
>>>
>>>
>>>
>>
>
>
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From bogdan at opensips.org  Mon Jul  4 10:45:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 11:45:08 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<CA+vx6KL+UhKLqLY3B1FDJ9evAgdDxYHMbHgE7h+_Enwy2Vh-6A@mail.gmail.com>
	<CA+vx6K+3S9FhgxQBee3JsVTkew2BZCns+WZzDEf_ZRZyzXS5kA@mail.gmail.com>
	<CA+vx6KLLsSgE=HOcZ3oXeMRFJvcJQ=q-btBwz82J+DG8bMqkCw@mail.gmail.com>
Message-ID: <577A2214.2040907@opensips.org>

Hi,

0. The cfg block you mentioned as removed does not exists in the cfg as 
per tutorial.

1. the "limit" column does not exist in the sipusers as per tutorial, so 
it might have been added in newer asterisk versions; not sure what is 
its meaning, but if setting it to 1 makes asterisk happy, it should be fine.

2. with odbc_voicemail, you do not have to add the users in 
voicemail.conf . Are you sure you properly set in 
/etc/asterisk/extconfig.conf the lines with:
        voicemail => odbc,asteriskcfg,vmusers
            voicemail => odbc,asteriskcfg,vmaliases

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 03.07.2016 22:59, Nabeel wrote:
> The last error message has been solved by removing the following lines 
> from opensips.cfg:
>
>        if (!db_does_uri_exist()) {
>                            send_reply("420","Bad Extension");
>                            exit;
>                    }
>
>                  t_newtran();
>                    t_reply("480", "Temporarily Unavailable");
>
>
>
> Now voicemail seems to be working, but only if manually adding users 
> to the voicemail.conf file. So the following questions remain:
>
> 1. Adding "limit = 1" to sipusers mysql view resolves a database 
> error. Is this the correct way to fix the error for voicemail integration?
>
> 2. How can voicemail.conf file be configured to use variable 
> substitution for all users in the OpenSIPS subscriber table?
>
> Nabeel
>
> On 2 July 2016 at 13:41, Nabeel <nabeelshikder at gmail.com 
> <mailto:nabeelshikder at gmail.com>> wrote:
>
>     In the latest version of Asterisk, there is a new file
>     voicemail.conf which must be configured correctly for voicemail,
>     but the tutorial does not mention this file at all. Please let me
>     know how to configure this file for integration with OpenSIPS.
>
>     Nabeel
>
>

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From bogdan at opensips.org  Mon Jul  4 11:13:34 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 12:13:34 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
Message-ID: <577A28BE.1010409@opensips.org>

Thank you Nabeel,

The number you added in voicemail file - does it exist in the 
sipuser/subscriber table ??

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 04.07.2016 11:38, Nabeel wrote:
>
> Hi Bogdan,
>
> I just added the column to the view by adding "NULL AS 
> `callbackextension`" to the SQL view definition. I haven't linked the 
> column to the subscriber column, so this may not be the correct 
> definition. However, it got rid of the error.
>
> About the voicemail.conf file, when I attempted to leave a voicemail 
> to a number in subscriber table, I got an error telling me that the 
> user/mailbox does not exist in voicemail.conf file. After adding that 
> number manually to the file, voicemail was working for that number. I 
> think this may also mean that the integration is not working correctly 
> in my current setup.
>
> Nabeel
>
> On 4 Jul 2016 8:48 a.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     Hi,
>
>     What is the definition you used for this new column ?
>
>     Best regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 02.07.2016 05:29, Nabeel wrote:
>>     In the last error message,/'//callbackextension = ?' /suggested
>>     that this column is missing from the sipusers mysql view. So I
>>     added this column to the view and now that error has been
>>     resolved. Only the following error remains now:
>>
>>         [Jul  2 03:25:48] WARNING[19330][C-00000005]: app.c:1633
>>         __ast_play_and_record: No audio available on
>>         SIP/domain.com-00000005??
>>
>>
>>
>>
>>
>>
>>     On 2 July 2016 at 02:36, Nabeel <nabeelshikder at gmail.com
>>     <mailto:nabeelshikder at gmail.com>> wrote:
>>
>>         The issue in my last Email has solved the error about missing
>>         extension. Now the following errors remain:
>>
>>             [Jul  2 02:29:18] WARNING[18226][C-00000001]:
>>             res_config_odbc.c:117 custom_prepare: SQL Prepare
>>             failed![SELECT * FROM sipusers WHERE host = ? AND
>>             callbackextension = ? AND port = ?]
>>             [Jul  2 02:29:22] WARNING[18269][C-00000000]: app.c:1633
>>             __ast_play_and_record: No audio available on
>>             SIP/domain.com-00000000??
>>
>>
>>
>>         On 2 July 2016 at 02:23, Nabeel <nabeelshikder at gmail.com
>>         <mailto:nabeelshikder at gmail.com>> wrote:
>>
>>             The tutorial contains a mistake where the priority
>>             ordering in extensions.conf should start with 1, not n:
>>
>>                 ; Voicemail
>>                 exten => _VMR_.,1,Ringing
>>                 exten => _VMR_.,n,Wait(1)
>>                 exten => _VMR_.,n,Answer
>>                 exten => _VMR_.,n,Wait(1)
>>                 exten => _VMR_.,n,Voicemail(${EXTEN:4}|u)
>>                 exten => _VMR_.,n,Hangup 
>>
>>                 ; Allow users to call their Voicemail directly
>>                 exten => VM_pickup,1,Ringing
>>                 exten => VM_pickup,n,wait(1)
>>                 exten => VM_pickup,n,VoicemailMain(${CALLERIDNUM}|s)
>>                 exten => VM_pickup,n,Hangup
>>
>>
>>
>>
>>
>

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From nabeelshikder at gmail.com  Mon Jul  4 11:54:31 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Mon, 4 Jul 2016 10:54:31 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <577A28BE.1010409@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
Message-ID: <CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>

0.  The line block was in the default OpenSIPS config, but I agree that it
is not in the tutorial so should be removed (for voicemail).

1. I think there is a misunderstanding here.  'limit' is not a column; I am
referring to the mysql LIMIT clause:

https://dev.mysql.com/doc/refman/5.5/en/select.html

2. In the tutorial, the following is added to extconfig.cfg:

sipusers => odbc,asterisk,sipusers sippeers => odbc,asterisk,sipusers
voicemail => odbc,asterisk,vmusers meetme => odbc,asterisk,meetme

However, this is different to the config in your last Email with
'asteriskcfg'. Please clarify.

For the phone number +447479189410 in the subscriber table, I had to add
the following line to voicemail.conf for it to work (notice the " |u " ):

+447479189410|u => 1234,Example Mailbox,root at localhost

Asterisk seems to look for the " |u " suffix in addition to the  phone
number, instead of just the phone number which is in the subscriber table.

Nabeel
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From bogdan at opensips.org  Mon Jul  4 11:54:35 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 12:54:35 +0300
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
	<CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
Message-ID: <577A325B.1070809@opensips.org>

Thank you Tito,

It looks like a crash in the memory manager, while trying to allocate a 
new structure. Do debug such problems there is no other way than 
enabling debugging for the memory manager (QM_MALLOC + DBG_MALLOC flags) 
- is this a production system with considerable load on it ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 01.07.2016 18:44, Tito Cumpen wrote:
> Bogdan,
>
>
> Correction I was using :
>
> version: opensips 2.2.0 (x86_64/linux)
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>
> git revision: d835721
>
> main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5
>
>
>
>
>
> On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen <tito at xsvoce.com 
> <mailto:tito at xsvoce.com>> wrote:
>
>     Bogdan,
>
>     Here is the backtrace:
>
>
>     Reading symbols from /usr/sbin/opensips...done.
>
>     [New LWP 2648]
>
>     [Thread debugging using libthread_db enabled]
>
>     Using host libthread_db library "/lib64/libthread_db.so.1".
>
>     Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u
>     opensips -g opensips -M 1024 -f /etc'.
>
>     Program terminated with signal 11, Segmentation fault.
>
>     #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>     qm=0x7fadc0809010) at mem/f_malloc.c:187
>
>     187*pf=n->u.nxt_free;
>
>     Missing separate debuginfos, use: debuginfo-install
>     cyrus-sasl-lib-2.1.26-20.el7_2.x86_64
>     glibc-2.17-106.el7_2.4.x86_64 gmp-6.0.0-12.el7_1.x86_64
>     gnutls-3.3.8-14.el7_2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64
>     krb5-libs-1.13.2-12.el7_2.x86_64 libcom_err-1.42.9-7.el7.x86_64
>     libcurl-7.29.0-25.el7.centos.x86_64 libffi-3.0.13-16.el7.x86_64
>     libgcc-4.8.5-4.el7.x86_64 libgcrypt-1.5.3-12.el7_1.1.x86_64
>     libgpg-error-1.12-3.el7.x86_64 libidn-1.28-4.el7.x86_64
>     libmicrohttpd-0.9.33-2.el7.x86_64 librabbitmq-0.5.2-1.el7.x86_64
>     libselinux-2.2.2-6.el7.x86_64 libssh2-1.4.3-10.el7_2.1.x86_64
>     libstdc++-4.8.5-4.el7.x86_64 libtasn1-3.8-2.el7.x86_64
>     nettle-2.7.1-4.el7.x86_64 nspr-4.11.0-1.el7_2.x86_64
>     nss-3.21.0-9.el7_2.x86_64
>     nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
>     nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
>     openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
>     pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
>     xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64
>
>     (gdb) bt full
>
>     #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>     qm=0x7fadc0809010) at mem/f_malloc.c:187
>
>             pf = 0x0
>
>             hash = 2051
>
>     #1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
>     mem/f_malloc.c:415
>
>             frag = 0x7fadc0898210
>
>             n = <optimized out>
>
>             hash = 2051
>
>             __FUNCTION__ = "fm_malloc"
>
>     #2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
>     ../proto_ws/ws_common.h:576
>
>             size = <optimized out>
>
>             req = <optimized out>
>
>             ret_code = WS_ERR_NONE
>
>             bk = <optimized out>
>
>             msg_len = <optimized out>
>
>             local_rcv = {src_ip = {af = 3082146304 <tel:3082146304>,
>     len = 32685, u = {addrl = {34359801655, 140720703251544}, addr32 =
>     {63287, 8, 394765400, 32764}, addr16 = {63287, 0, 8, 0, 42072,
>     6023, 32764, 0},
>
>                   addr =
>     "7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}}, dst_ip =
>     {af = 3080002780, len = 32685, u = {addrl = {4, 63119970000},
>     addr32 = {4, 0, 2990427856, 14}, addr16 = {4, 0, 0,
>
>                     0, 20176, 45630, 14, 0}, addr =
>     "\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}},
>     src_port = 52976, dst_port = 49287, proto = 32685, proto_reserved1
>     = -1214962633,
>
>               proto_reserved2 = 32685, src_su = {s = {sa_family = 2,
>     sa_data = ")\350n\343\276C\360#%\275\016\000\000"}, sin =
>     {sin_family = 2, sin_port = 59433, sin_addr = {s_addr = 1136583534},
>
>                   sin_zero = "\360#%\275\016\000\000"}, sin6 =
>     {sin6_family = 2, sin6_port = 59433, sin6_flowinfo = 1136583534,
>     sin6_addr = {__in6_u = {
>
>                       __u6_addr8 =
>     "\360#%\275\016\000\000\000\360?\300\255\177\000", __u6_addr16 =
>     {9200, 48421, 14, 0, 52976, 49287, 32685, 0}, __u6_addr32 =
>     {3173327856, 14, 3230125808, 32685}}},
>
>                   sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}
>
>             newreq = <optimized out>
>
>             msg_buf = <optimized out>
>
>     #3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized out>)
>     at proto_wss.c:428
>
>             size = <optimized out>
>
>             __FUNCTION__ = "wss_read_req"
>
>     #4  0x000000000059c374 in handle_io (fm=<optimized out>,
>     idx=idx at entry=0, event_type=event_type at entry=1) at
>     net/net_tcp_proc.c:205
>
>             ret = 0
>
>             n = <optimized out>
>
>             con = 0x7fadbeb3f718
>
>             s = 57
>
>             rw = <optimized out>
>
>             resp = <optimized out>
>
>             response = {140384203757736, 1}
>
>             __FUNCTION__ = "handle_io"
>
>     #5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized out>,
>     t=<optimized out>, repeat=<optimized out>) at
>     net/../io_wait_loop.h:221
>
>             ret = <optimized out>
>
>             e = <optimized out>
>
>             n = 1
>
>             r = 0
>
>     #6  tcp_worker_proc (unix_sock=<optimized out>) at
>     net/net_tcp_proc.c:312
>
>             __FUNCTION__ = "tcp_worker_proc"
>
>     #7  0x00000000005a7fc9 in tcp_start_processes
>     (chd_rank=chd_rank at entry=0x8417e0 <chd_rank.11018>,
>     startup_done=startup_done at entry=0x7fadbe978738) at net/net_tcp.c:1761
>
>             r = 5
>
>             reader_fd = {50, 52}
>
>             pid = 0
>
>             load_p = 0x7fadbe979398
>
>             __FUNCTION__ = "tcp_start_processes"
>
>     #8  0x0000000000419ef5 in main_loop () at main.c:677
>
>             startup_done = 0x7fadbe978738
>
>             chd_rank = 18
>
>     #9  main (argc=<optimized out>, argv=<optimized out>) at main.c:1249
>
>             cfg_stream = <optimized out>
>
>             c = <optimized out>
>
>             r = <optimized out>
>
>             tmp = 0x7ffc1787bf65 ""
>
>             tmp_len = <optimized out>
>
>             port = <optimized out>
>
>             proto = <optimized out>
>
>             protos_no = <optimized out>
>
>             options = 0x5dae60
>     "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"
>
>     ---Type <return> to continue, or q <return> to quit---
>
>             ret = -1
>
>             seed = 995413301
>
>             rfd = <optimized out>
>
>             __FUNCTION__ = "main"
>
>
>
>     currently using version: opensips 2.3.0-dev (x86_64/linux)
>
>     flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>     QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
>     ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
>     16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>
>     poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>
>     git revision: 92434ef
>
>     main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>
>
>     On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu
>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>         Hi Tito,
>
>         If opensips crashes, were you able to extract a backtrace from
>         the core file(s) ?
>
>         Thanks and regards,
>
>         Bogdan-Andrei Iancu
>         OpenSIPS Founder and Developer
>         http://www.opensips-solutions.com
>
>         On 27.06.2016 21:20, Tito Cumpen wrote:
>>         Group,
>>
>>
>>         I am experiencing strange behavior when configuring sip tls
>>         on port 443. At time opensips crashes or stops accepting new
>>         connections. Here are the tcp configs I am using:
>>
>>         #disable_tcp=no
>>
>>         tcp_connection_lifetime=3600
>>
>>         tcp_connect_timeout=3
>>
>>         tcp_keepidle = 30
>>
>>         tcp_keepinterval = 5
>>
>>         tcp_keepalive = 1
>>
>>         tcp_keepcount = 5
>>
>>         tcp_max_msg_time = 8
>>
>>         tcp_children=10
>>
>>
>>         Any idea what would case this? I am assuming there are probes
>>         out in the internet that eventually make opensips crash?
>>
>>         Thanks,
>>         Tito
>>
>>
>>         _______________________________________________
>>         Users mailing list
>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>

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From bogdan at opensips.org  Mon Jul  4 13:03:38 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 14:03:38 +0300
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
Message-ID: <577A428A.6020505@opensips.org>

Hi Alain,

Thanks for the trace, I managed to find out the problem. I can provide 
you a fix to test - do you have OpenSIPS compiled from sources ? or ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 01.07.2016 16:56, Alain Bieuzent wrote:
>
> Hi Bogdan,
>
> As i see in trace (see pcap attach) the first sip OTIONS is send with 
> the correct source IP (..........) to my 2 gateway (see frame 3 and 4) 
> and later there is 2 other sip OPTIONS sent with my private IP 
> (............) (see frame 10 and 11). But all the packet are sending 
> from the same interface (00:50:56:9a:78:93).
>
> My network configuration :
>
> [....]
>
> ######################################################################
>
> ## Connectivity
>
> ######################################################################
>
> [....]
>
> Thanks.
>
> *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date : *vendredi 1 juillet 2016 12:04
> *? : *OpenSIPS users mailling list <users at lists.opensips.org>, Alain 
> Bieuzent <alain.bieuzent at free.fr>
> *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load 
> balancer Probing problem
>
> Hello Alain,
>
> You mean when comes to sent to ping a single destination, there are 
> actually 2 OPTIONS sent out (on different interfaces) or it alternates 
> (one ping is sent via eth0, the next one via eth1) ?
>
> if you have a pcap to show the problem, please send it to me off-list.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 30.06.2016 12:49, Alain Bieuzent wrote:
>
>     Hi  all,
>
>     I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.
>
>     My server has 3 interfaces:
>
>     eth0 with public IP
>
>     eth0:0 with public IP provide via a keepalived script
>
>     eth1 with a private IP
>
>     Opensips listen on IP of eth0:0 and on eth1
>
>     Mhomed=no
>
>     Since i migrate to V2.1.3, load balancer probing is UP and down.
>     After checking traffic, i find for each gateway used by load
>     balancer table, there is 2 SIP OPTIONS sent.
>
>     -First is sent with source IP of eth0:0
>
>     -Second is sent with source IP of eth1
>
>     Any idea ?
>
>     Thanls
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From alain.bieuzent at free.fr  Mon Jul  4 14:34:15 2016
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Mon, 04 Jul 2016 14:34:15 +0200
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <577A428A.6020505@opensips.org>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
	<577A428A.6020505@opensips.org>
Message-ID: <9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>

Hi Bogdan,

 

Yes i compiled form source.

 

Regards

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : lundi 4 juillet 2016 13:03
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Thanks for the trace, I managed to find out the problem. I can provide you a fix to test - do you have OpenSIPS compiled from sources ? or ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.07.2016 16:56, Alain Bieuzent wrote:

Hi Bogdan,

 

As i see in trace (see pcap attach) the first sip OTIONS is send with the correct source IP (..........) to my 2 gateway (see frame 3 and 4) and later there is 2 other sip OPTIONS sent with my private IP (............) (see frame 10 and 11). But all the packet are sending from the same interface (00:50:56:9a:78:93).

 

My network configuration :

 

[....] 

 

 

######################################################################

## Connectivity

######################################################################

[....] 

 

 

Thanks.

 

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : vendredi 1 juillet 2016 12:04
? : OpenSIPS users mailling list <users at lists.opensips.org>, Alain Bieuzent <alain.bieuzent at free.fr>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hello Alain,

You mean when comes to sent to ping a single destination, there are actually 2 OPTIONS sent out (on different interfaces) or it alternates (one ping is sent via eth0, the next one via eth1) ?

if you have a pcap to show the problem, please send it to me off-list.

Best regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 30.06.2016 12:49, Alain Bieuzent wrote:

Hi  all,

 

I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.

 

My server has 3 interfaces:

eth0 with public IP

eth0:0 with public IP provide via a keepalived script

eth1 with a private IP

 

Opensips listen on IP of eth0:0 and on eth1

Mhomed=no

 

 

Since i migrate to V2.1.3, load balancer probing is UP and down. After checking traffic, i find for each gateway used by load balancer table, there is 2 SIP OPTIONS sent.

-          First is sent with source IP of eth0:0

-          Second is sent with source IP of eth1

 

Any idea ?

 

Thanls





_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 

 

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From bogdan at opensips.org  Mon Jul  4 19:08:28 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 4 Jul 2016 20:08:28 +0300
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
	<577A428A.6020505@opensips.org>
	<9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>
Message-ID: <577A980C.9010706@opensips.org>

Hi Alain,

Could you please test the following patch.

Thank and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 04.07.2016 15:34, Alain Bieuzent wrote:
>
> Hi Bogdan,
>
> Yes i compiled form source.
>
> Regards
>
> *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date : *lundi 4 juillet 2016 13:03
> *? : *Alain Bieuzent <alain.bieuzent at free.fr>, 
> "users at lists.opensips.org" <users at lists.opensips.org>
> *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load 
> balancer Probing problem
>
> Hi Alain,
>
> Thanks for the trace, I managed to find out the problem. I can provide 
> you a fix to test - do you have OpenSIPS compiled from sources ? or ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 01.07.2016 16:56, Alain Bieuzent wrote:
>
>     Hi Bogdan,
>
>     As i see in trace (see pcap attach) the first sip OTIONS is send
>     with the correct source IP (..........) to my 2 gateway (see frame
>     3 and 4) and later there is 2 other sip OPTIONS sent with my
>     private IP (............) (see frame 10 and 11). But all the
>     packet are sending from the same interface (00:50:56:9a:78:93).
>
>     My network configuration :
>
>     [....]
>
>     ######################################################################
>
>     ## Connectivity
>
>     ######################################################################
>
>     [....]
>
>     Thanks.
>
>     *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>
>     *Date : *vendredi 1 juillet 2016 12:04
>     *? : *OpenSIPS users mailling list <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>, Alain Bieuzent
>     <alain.bieuzent at free.fr> <mailto:alain.bieuzent at free.fr>
>     *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load
>     balancer Probing problem
>
>     Hello Alain,
>
>     You mean when comes to sent to ping a single destination, there
>     are actually 2 OPTIONS sent out (on different interfaces) or it
>     alternates (one ping is sent via eth0, the next one via eth1) ?
>
>     if you have a pcap to show the problem, please send it to me off-list.
>
>     Best regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 30.06.2016 12:49, Alain Bieuzent wrote:
>
>         Hi  all,
>
>         I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.
>
>         My server has 3 interfaces:
>
>         eth0 with public IP
>
>         eth0:0 with public IP provide via a keepalived script
>
>         eth1 with a private IP
>
>         Opensips listen on IP of eth0:0 and on eth1
>
>         Mhomed=no
>
>         Since i migrate to V2.1.3, load balancer probing is UP and
>         down. After checking traffic, i find for each gateway used by
>         load balancer table, there is 2 SIP OPTIONS sent.
>
>         -First is sent with source IP of eth0:0
>
>         -Second is sent with source IP of eth1
>
>         Any idea ?
>
>         Thanls
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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diff --git a/forward.c b/forward.c
index 16f40ae..d74a521 100644
--- a/forward.c
+++ b/forward.c
@@ -223,12 +223,10 @@ struct socket_info* get_send_socket(struct sip_msg *msg,
 	 * eg: ipv4 -> ipv6 or ipv6 -> ipv4) */
 	switch(proto){
 		case PROTO_UDP:
-			if ((bind_address)&&(to->s.sa_family==bind_address->address.af)&&
-			(bind_address->proto==PROTO_UDP)) {
-				send_sock = (msg &&
-				msg->rcv.bind_address->address.af==bind_address->address.af &&
-				msg->rcv.bind_address->proto==bind_address->proto) ?
-					msg->rcv.bind_address : bind_address;
+			if (msg &&
+			msg->rcv.bind_address->address.af==bind_address->address.af &&
+			msg->rcv.bind_address->proto==bind_address->proto) {
+				send_sock = msg->rcv.bind_address;
 				break;
 			}
 			/* default logic for all protos */

From nabeelshikder at gmail.com  Tue Jul  5 07:54:32 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Tue, 5 Jul 2016 06:54:32 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
Message-ID: <CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>

I have been able to solve the issue of loading numbers without using the
voicemail.conf file.

After adding the line *'voicemail => odbc,asterisk,vmaliases'* to
extconfig.cfg, I removed the suffix " |u " from extensions.conf:

exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)



Now all phone numbers in the subscriber table are correctly linked to their
mailboxes and I can successfully retrieve voice messages.

I have set 'odbcstorage=asterisk' and 'odbctable=voicemessages' in
voicemail.conf, but the 'voicemessages' table remains empty after leaving a
message.

Nabeel
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From alain.bieuzent at free.fr  Tue Jul  5 09:24:40 2016
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 05 Jul 2016 09:24:40 +0200
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <577A980C.9010706@opensips.org>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
	<577A428A.6020505@opensips.org>
	<9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>
	<577A980C.9010706@opensips.org>
Message-ID: <DC35B9D4-4F9F-482C-8EE1-51297E7FE4F0@free.fr>

Ho bogdan,

 

Tested your patch this morning, it works.

 

Many thanks.

 

Regards

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : lundi 4 juillet 2016 19:08
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Could you please test the following patch.

Thank and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04.07.2016 15:34, Alain Bieuzent wrote:

Hi Bogdan,

 

Yes i compiled form source.

 

Regards

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : lundi 4 juillet 2016 13:03
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Thanks for the trace, I managed to find out the problem. I can provide you a fix to test - do you have OpenSIPS compiled from sources ? or ?

Regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.07.2016 16:56, Alain Bieuzent wrote:

Hi Bogdan,

 

As i see in trace (see pcap attach) the first sip OTIONS is send with the correct source IP (..........) to my 2 gateway (see frame 3 and 4) and later there is 2 other sip OPTIONS sent with my private IP (............) (see frame 10 and 11). But all the packet are sending from the same interface (00:50:56:9a:78:93).

 

My network configuration :

 

[....] 

 

 

######################################################################

## Connectivity

######################################################################

[....] 

 

 

Thanks.

 

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : vendredi 1 juillet 2016 12:04
? : OpenSIPS users mailling list <users at lists.opensips.org>, Alain Bieuzent <alain.bieuzent at free.fr>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hello Alain,

You mean when comes to sent to ping a single destination, there are actually 2 OPTIONS sent out (on different interfaces) or it alternates (one ping is sent via eth0, the next one via eth1) ?

if you have a pcap to show the problem, please send it to me off-list.

Best regards,



Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 30.06.2016 12:49, Alain Bieuzent wrote:

Hi  all,

 

I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.

 

My server has 3 interfaces:

eth0 with public IP

eth0:0 with public IP provide via a keepalived script

eth1 with a private IP

 

Opensips listen on IP of eth0:0 and on eth1

Mhomed=no

 

 

Since i migrate to V2.1.3, load balancer probing is UP and down. After checking traffic, i find for each gateway used by load balancer table, there is 2 SIP OPTIONS sent.

-          First is sent with source IP of eth0:0

-          Second is sent with source IP of eth1

 

Any idea ?

 

Thanls






_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 

 

 

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From bogdan at opensips.org  Tue Jul  5 11:21:11 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 5 Jul 2016 12:21:11 +0300
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <DC35B9D4-4F9F-482C-8EE1-51297E7FE4F0@free.fr>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
	<577A428A.6020505@opensips.org>
	<9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>
	<577A980C.9010706@opensips.org>
	<DC35B9D4-4F9F-482C-8EE1-51297E7FE4F0@free.fr>
Message-ID: <577B7C07.5090501@opensips.org>

Hi Alain,

Thanks for your testing and feedback. We found some issues with the 
patch I sent you yesterday, so please consider for usage this second one 
(revert the first one please).

Also, I will push this fix on opensips trunk, to get it tested and to be 
sure there are no side effects.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 10:24, Alain Bieuzent wrote:
>
> Ho bogdan,
>
> Tested your patch this morning, it works.
>
> Many thanks.
>
> Regards
>
> *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date : *lundi 4 juillet 2016 19:08
> *? : *Alain Bieuzent <alain.bieuzent at free.fr>, 
> "users at lists.opensips.org" <users at lists.opensips.org>
> *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load 
> balancer Probing problem
>
> Hi Alain,
>
> Could you please test the following patch.
>
> Thank and regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 04.07.2016 15:34, Alain Bieuzent wrote:
>
>     Hi Bogdan,
>
>     Yes i compiled form source.
>
>     Regards
>
>     *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
>     <mailto:bogdan at opensips.org>
>     *Date : *lundi 4 juillet 2016 13:03
>     *? : *Alain Bieuzent <alain.bieuzent at free.fr>
>     <mailto:alain.bieuzent at free.fr>, "users at lists.opensips.org"
>     <mailto:users at lists.opensips.org> <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>
>     *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load
>     balancer Probing problem
>
>     Hi Alain,
>
>     Thanks for the trace, I managed to find out the problem. I can
>     provide you a fix to test - do you have OpenSIPS compiled from
>     sources ? or ?
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 01.07.2016 16:56, Alain Bieuzent wrote:
>
>         Hi Bogdan,
>
>         As i see in trace (see pcap attach) the first sip OTIONS is
>         send with the correct source IP (..........) to my 2 gateway
>         (see frame 3 and 4) and later there is 2 other sip OPTIONS
>         sent with my private IP (............) (see frame 10 and 11).
>         But all the packet are sending from the same interface
>         (00:50:56:9a:78:93).
>
>         My network configuration :
>
>         [....]
>
>         ######################################################################
>
>         ## Connectivity
>
>         ######################################################################
>
>         [....]
>
>         Thanks.
>
>         *De : *Bogdan-Andrei Iancu <bogdan at opensips.org>
>         <mailto:bogdan at opensips.org>
>         *Date : *vendredi 1 juillet 2016 12:04
>         *? : *OpenSIPS users mailling list <users at lists.opensips.org>
>         <mailto:users at lists.opensips.org>, Alain Bieuzent
>         <alain.bieuzent at free.fr> <mailto:alain.bieuzent at free.fr>
>         *Objet : *Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load
>         balancer Probing problem
>
>         Hello Alain,
>
>         You mean when comes to sent to ping a single destination,
>         there are actually 2 OPTIONS sent out (on different
>         interfaces) or it alternates (one ping is sent via eth0, the
>         next one via eth1) ?
>
>         if you have a pcap to show the problem, please send it to me
>         off-list.
>
>         Best regards,
>
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 30.06.2016 12:49, Alain Bieuzent wrote:
>
>             Hi all,
>
>             I?m currently trying to migrate my opensips V1.11.5 to 
>             V2.1.3.
>
>             My server has 3 interfaces:
>
>             eth0 with public IP
>
>             eth0:0 with public IP provide via a keepalived script
>
>             eth1 with a private IP
>
>             Opensips listen on IP of eth0:0 and on eth1
>
>             Mhomed=no
>
>             Since i migrate to V2.1.3, load balancer probing is UP and
>             down. After checking traffic, i find for each gateway used
>             by load balancer table, there is 2 SIP OPTIONS sent.
>
>             -First is sent with source IP of eth0:0
>
>             -Second is sent with source IP of eth1
>
>             Any idea ?
>
>             Thanls
>
>
>
>
>
>
>             _______________________________________________
>
>             Users mailing list
>
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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-------------- next part --------------
diff --git a/forward.c b/forward.c
index 16f40ae..22156f3 100644
--- a/forward.c
+++ b/forward.c
@@ -223,12 +223,10 @@ struct socket_info* get_send_socket(struct sip_msg *msg,
 	 * eg: ipv4 -> ipv6 or ipv6 -> ipv4) */
 	switch(proto){
 		case PROTO_UDP:
-			if ((bind_address)&&(to->s.sa_family==bind_address->address.af)&&
-			(bind_address->proto==PROTO_UDP)) {
-				send_sock = (msg &&
-				msg->rcv.bind_address->address.af==bind_address->address.af &&
-				msg->rcv.bind_address->proto==bind_address->proto) ?
-					msg->rcv.bind_address : bind_address;
+			if (msg &&
+			msg->rcv.bind_address->address.af==to->s.sa_family &&
+			msg->rcv.bind_address->proto==PROTO_UDP) {
+				send_sock = msg->rcv.bind_address;
 				break;
 			}
 			/* default logic for all protos */

From tito at xsvoce.com  Tue Jul  5 13:37:34 2016
From: tito at xsvoce.com (Tito Cumpen)
Date: Tue, 5 Jul 2016 07:37:34 -0400
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <577A325B.1070809@opensips.org>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
	<CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
	<577A325B.1070809@opensips.org>
Message-ID: <CANZPVB4MMu5N=OuTEVg6vqTksOmcKVRmx4nurei3Fgs9MuW_bg@mail.gmail.com>

Bogdan,

Here is the backtrace with those flags enabled

gdb /usr/sbin/opensips core.2685

GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-80.el7

Copyright (C) 2013 Free Software Foundation, Inc.

License GPLv3+: GNU GPL version 3 or later <http://gnu.org/licenses/gpl.html
>

This is free software: you are free to change and redistribute it.

There is NO WARRANTY, to the extent permitted by law.  Type "show copying"

and "show warranty" for details.

This GDB was configured as "x86_64-redhat-linux-gnu".

For bug reporting instructions, please see:

<http://www.gnu.org/software/gdb/bugs/>...

Reading symbols from /usr/sbin/opensips...done.

[New LWP 2685]

[Thread debugging using libthread_db enabled]

Using host libthread_db library "/lib64/libthread_db.so.1".

Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips
-g opensips -M 1024 -f /etc'.

Program terminated with signal 6, Aborted.

#0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at
../nptl/sysdeps/unix/sysv/linux/raise.c:56

56   return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig);

(gdb)


Let me know if you need anything else



On Mon, Jul 4, 2016 at 5:54 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Thank you Tito,
>
> It looks like a crash in the memory manager, while trying to allocate a
> new structure. Do debug such problems there is no other way than enabling
> debugging for the memory manager (QM_MALLOC + DBG_MALLOC flags) - is this a
> production system with considerable load on it ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 01.07.2016 18:44, Tito Cumpen wrote:
>
> Bogdan,
>
>
> Correction I was using :
>
> version: opensips 2.2.0 (x86_64/linux)
>
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
>
> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>
> git revision: d835721
>
> main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5
>
>
>
>
>
> On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen <tito at xsvoce.com> wrote:
>
>> Bogdan,
>>
>> Here is the backtrace:
>>
>>
>> Reading symbols from /usr/sbin/opensips...done.
>>
>> [New LWP 2648]
>>
>> [Thread debugging using libthread_db enabled]
>>
>> Using host libthread_db library "/lib64/libthread_db.so.1".
>>
>> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u
>> opensips -g opensips -M 1024 -f /etc'.
>>
>> Program terminated with signal 11, Segmentation fault.
>>
>> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>> qm=0x7fadc0809010) at mem/f_malloc.c:187
>>
>> 187 *pf=n->u.nxt_free;
>>
>> Missing separate debuginfos, use: debuginfo-install
>> cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 glibc-2.17-106.el7_2.4.x86_64
>> gmp-6.0.0-12.el7_1.x86_64 gnutls-3.3.8-14.el7_2.x86_64
>> keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.13.2-12.el7_2.x86_64
>> libcom_err-1.42.9-7.el7.x86_64 libcurl-7.29.0-25.el7.centos.x86_64
>> libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64
>> libgcrypt-1.5.3-12.el7_1.1.x86_64 libgpg-error-1.12-3.el7.x86_64
>> libidn-1.28-4.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64
>> librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64
>> libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64
>> libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64
>> nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64
>> nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
>> nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
>> openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
>> pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
>> xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64
>>
>> (gdb) bt full
>>
>> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>> qm=0x7fadc0809010) at mem/f_malloc.c:187
>>
>>         pf = 0x0
>>
>>         hash = 2051
>>
>> #1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
>> mem/f_malloc.c:415
>>
>>         frag = 0x7fadc0898210
>>
>>         n = <optimized out>
>>
>>         hash = 2051
>>
>>         __FUNCTION__ = "fm_malloc"
>>
>> #2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
>> ../proto_ws/ws_common.h:576
>>
>>         size = <optimized out>
>>
>>         req = <optimized out>
>>
>>         ret_code = WS_ERR_NONE
>>
>>         bk = <optimized out>
>>
>>         msg_len = <optimized out>
>>
>>         local_rcv = {src_ip = {af = 3082146304, len = 32685, u = {addrl
>> = {34359801655, 140720703251544}, addr32 = {63287, 8, 394765400, 32764},
>> addr16 = {63287, 0, 8, 0, 42072, 6023, 32764, 0},
>>
>>               addr =
>> "7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}}, dst_ip = {af =
>> 3080002780, len = 32685, u = {addrl = {4, 63119970000}, addr32 = {4, 0,
>> 2990427856, 14}, addr16 = {4, 0, 0,
>>
>>                 0, 20176, 45630, 14, 0}, addr =
>> "\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}}, src_port =
>> 52976, dst_port = 49287, proto = 32685, proto_reserved1 = -1214962633,
>>
>>           proto_reserved2 = 32685, src_su = {s = {sa_family = 2, sa_data
>> = ")\350n\343\276C\360#%\275\016\000\000"}, sin = {sin_family = 2, sin_port
>> = 59433, sin_addr = {s_addr = 1136583534},
>>
>>               sin_zero = "\360#%\275\016\000\000"}, sin6 = {sin6_family =
>> 2, sin6_port = 59433, sin6_flowinfo = 1136583534, sin6_addr = {__in6_u = {
>>
>>                   __u6_addr8 =
>> "\360#%\275\016\000\000\000\360?\300\255\177\000", __u6_addr16 = {9200,
>> 48421, 14, 0, 52976, 49287, 32685, 0}, __u6_addr32 = {3173327856, 14,
>> 3230125808, 32685}}},
>>
>>               sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}
>>
>>         newreq = <optimized out>
>>
>>         msg_buf = <optimized out>
>>
>> #3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized out>) at
>> proto_wss.c:428
>>
>>         size = <optimized out>
>>
>>         __FUNCTION__ = "wss_read_req"
>>
>> #4  0x000000000059c374 in handle_io (fm=<optimized out>, idx=idx at entry=0,
>> event_type=event_type at entry=1) at net/net_tcp_proc.c:205
>>
>>         ret = 0
>>
>>         n = <optimized out>
>>
>>         con = 0x7fadbeb3f718
>>
>>         s = 57
>>
>>         rw = <optimized out>
>>
>>         resp = <optimized out>
>>
>>         response = {140384203757736, 1}
>>
>>         __FUNCTION__ = "handle_io"
>>
>> #5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized out>,
>> t=<optimized out>, repeat=<optimized out>) at net/../io_wait_loop.h:221
>>
>>         ret = <optimized out>
>>
>>         e = <optimized out>
>>
>>         n = 1
>>
>>         r = 0
>>
>> #6  tcp_worker_proc (unix_sock=<optimized out>) at net/net_tcp_proc.c:312
>>
>>         __FUNCTION__ = "tcp_worker_proc"
>>
>> #7  0x00000000005a7fc9 in tcp_start_processes (chd_rank=chd_rank at entry=0x8417e0
>> <chd_rank.11018>, startup_done=startup_done at entry=0x7fadbe978738) at
>> net/net_tcp.c:1761
>>
>>         r = 5
>>
>>         reader_fd = {50, 52}
>>
>>         pid = 0
>>
>>         load_p = 0x7fadbe979398
>>
>>         __FUNCTION__ = "tcp_start_processes"
>>
>> #8  0x0000000000419ef5 in main_loop () at main.c:677
>>
>>         startup_done = 0x7fadbe978738
>>
>>         chd_rank = 18
>>
>> #9  main (argc=<optimized out>, argv=<optimized out>) at main.c:1249
>>
>>         cfg_stream = <optimized out>
>>
>>         c = <optimized out>
>>
>>         r = <optimized out>
>>
>>         tmp = 0x7ffc1787bf65 ""
>>
>>         tmp_len = <optimized out>
>>
>>         port = <optimized out>
>>
>>         proto = <optimized out>
>>
>>         protos_no = <optimized out>
>>
>>         options = 0x5dae60 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"
>>
>> ---Type <return> to continue, or q <return> to quit---
>>
>>         ret = -1
>>
>>         seed = 995413301
>>
>>         rfd = <optimized out>
>>
>>         __FUNCTION__ = "main"
>>
>>
>>
>> currently using version: opensips 2.3.0-dev (x86_64/linux)
>>
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>
>> git revision: 92434ef
>>
>> main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>>
>> On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu <
>> <bogdan at opensips.org>bogdan at opensips.org> wrote:
>>
>>> Hi Tito,
>>>
>>> If opensips crashes, were you able to extract a backtrace from the core
>>> file(s) ?
>>>
>>> Thanks and regards,
>>>
>>> Bogdan-Andrei Iancu
>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>
>>> On 27.06.2016 21:20, Tito Cumpen wrote:
>>>
>>> Group,
>>>
>>>
>>> I am experiencing strange behavior when configuring sip tls on port 443.
>>> At time opensips crashes or stops accepting new connections. Here are the
>>> tcp configs I am using:
>>>
>>> #disable_tcp=no
>>>
>>> tcp_connection_lifetime=3600
>>>
>>> tcp_connect_timeout=3
>>>
>>> tcp_keepidle = 30
>>>
>>> tcp_keepinterval = 5
>>>
>>> tcp_keepalive = 1
>>>
>>> tcp_keepcount = 5
>>>
>>> tcp_max_msg_time = 8
>>> tcp_children=10
>>>
>>>
>>> Any idea what would case this? I am assuming there are probes out in the
>>> internet that eventually make opensips crash?
>>>
>>> Thanks,
>>> Tito
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>
>
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From bogdan at opensips.org  Tue Jul  5 13:58:00 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 5 Jul 2016 14:58:00 +0300
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <CANZPVB4MMu5N=OuTEVg6vqTksOmcKVRmx4nurei3Fgs9MuW_bg@mail.gmail.com>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
	<CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
	<577A325B.1070809@opensips.org>
	<CANZPVB4MMu5N=OuTEVg6vqTksOmcKVRmx4nurei3Fgs9MuW_bg@mail.gmail.com>
Message-ID: <577BA0C8.8000003@opensips.org>

Hi Tito,

Could you print the whole backtrace ?

Thanks,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 14:37, Tito Cumpen wrote:
> Bogdan,
>
> Here is the backtrace with those flags enabled
>
> gdb /usr/sbin/opensips core.2685
>
> GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-80.el7
>
> Copyright (C) 2013 Free Software Foundation, Inc.
>
> License GPLv3+: GNU GPL version 3 or later 
> <http://gnu.org/licenses/gpl.html>
>
> This is free software: you are free to change and redistribute it.
>
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
>
> and "show warranty" for details.
>
> This GDB was configured as "x86_64-redhat-linux-gnu".
>
> For bug reporting instructions, please see:
>
> <http://www.gnu.org/software/gdb/bugs/>...
>
> Reading symbols from /usr/sbin/opensips...done.
>
> [New LWP 2685]
>
> [Thread debugging using libthread_db enabled]
>
> Using host libthread_db library "/lib64/libthread_db.so.1".
>
> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u 
> opensips -g opensips -M 1024 -f /etc'.
>
> Program terminated with signal 6, Aborted.
>
> #0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at 
> ../nptl/sysdeps/unix/sysv/linux/raise.c:56
>
> 56  return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig);
>
> (gdb)
>
>
> Let me know if you need anything else
>
>
>
> On Mon, Jul 4, 2016 at 5:54 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Thank you Tito,
>
>     It looks like a crash in the memory manager, while trying to
>     allocate a new structure. Do debug such problems there is no other
>     way than enabling debugging for the memory manager (QM_MALLOC +
>     DBG_MALLOC flags) - is this a production system with considerable
>     load on it ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 01.07.2016 18:44, Tito Cumpen wrote:
>>     Bogdan,
>>
>>
>>     Correction I was using :
>>
>>     version: opensips 2.2.0 (x86_64/linux)
>>
>>     flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>     F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>>     ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
>>     16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>>     poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>
>>     git revision: d835721
>>
>>     main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5
>>
>>
>>
>>
>>
>>     On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen <tito at xsvoce.com
>>     <mailto:tito at xsvoce.com>> wrote:
>>
>>         Bogdan,
>>
>>         Here is the backtrace:
>>
>>
>>         Reading symbols from /usr/sbin/opensips...done.
>>
>>         [New LWP 2648]
>>
>>         [Thread debugging using libthread_db enabled]
>>
>>         Using host libthread_db library "/lib64/libthread_db.so.1".
>>
>>         Core was generated by `/sbin/opensips -P
>>         /var/run/opensips.pid -u opensips -g opensips -M 1024 -f /etc'.
>>
>>         Program terminated with signal 11, Segmentation fault.
>>
>>         #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>>         qm=0x7fadc0809010) at mem/f_malloc.c:187
>>
>>         187*pf=n->u.nxt_free;
>>
>>         Missing separate debuginfos, use: debuginfo-install
>>         cyrus-sasl-lib-2.1.26-20.el7_2.x86_64
>>         glibc-2.17-106.el7_2.4.x86_64 gmp-6.0.0-12.el7_1.x86_64
>>         gnutls-3.3.8-14.el7_2.x86_64 keyutils-libs-1.5.8-3.el7.x86_64
>>         krb5-libs-1.13.2-12.el7_2.x86_64
>>         libcom_err-1.42.9-7.el7.x86_64
>>         libcurl-7.29.0-25.el7.centos.x86_64
>>         libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64
>>         libgcrypt-1.5.3-12.el7_1.1.x86_64
>>         libgpg-error-1.12-3.el7.x86_64 libidn-1.28-4.el7.x86_64
>>         libmicrohttpd-0.9.33-2.el7.x86_64
>>         librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64
>>         libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64
>>         libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64
>>         nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64
>>         nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
>>         nss-util-3.21.0-2.2.el7_2.x86_64
>>         openldap-2.4.40-9.el7_2.x86_64
>>         openssl-libs-1.0.1e-51.el7_2.4.x86_64
>>         p11-kit-0.20.7-3.el7.x86_64 pcre-8.32-15.el7.x86_64
>>         trousers-0.3.13-1.el7.x86_64 xz-libs-5.1.2-12alpha.el7.x86_64
>>         zlib-1.2.7-15.el7.x86_64
>>
>>         (gdb) bt full
>>
>>         #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>>         qm=0x7fadc0809010) at mem/f_malloc.c:187
>>
>>                 pf = 0x0
>>
>>                 hash = 2051
>>
>>         #1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
>>         mem/f_malloc.c:415
>>
>>                 frag = 0x7fadc0898210
>>
>>                 n = <optimized out>
>>
>>                 hash = 2051
>>
>>                 __FUNCTION__ = "fm_malloc"
>>
>>         #2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
>>         ../proto_ws/ws_common.h:576
>>
>>                 size = <optimized out>
>>
>>                 req = <optimized out>
>>
>>                 ret_code = WS_ERR_NONE
>>
>>                 bk = <optimized out>
>>
>>                 msg_len = <optimized out>
>>
>>                 local_rcv = {src_ip = {af = 3082146304
>>         <tel:3082146304>, len = 32685, u = {addrl = {34359801655,
>>         140720703251544}, addr32 = {63287, 8, 394765400, 32764},
>>         addr16 = {63287, 0, 8, 0, 42072, 6023, 32764, 0},
>>
>>                       addr =
>>         "7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}},
>>         dst_ip = {af = 3080002780, len = 32685, u = {addrl = {4,
>>         63119970000}, addr32 = {4, 0, 2990427856, 14}, addr16 = {4,
>>         0, 0,
>>
>>                         0, 20176, 45630, 14, 0}, addr =
>>         "\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}},
>>         src_port = 52976, dst_port = 49287, proto = 32685,
>>         proto_reserved1 = -1214962633,
>>
>>                   proto_reserved2 = 32685, src_su = {s = {sa_family =
>>         2, sa_data = ")\350n\343\276C\360#%\275\016\000\000"}, sin =
>>         {sin_family = 2, sin_port = 59433, sin_addr = {s_addr =
>>         1136583534},
>>
>>                       sin_zero = "\360#%\275\016\000\000"}, sin6 =
>>         {sin6_family = 2, sin6_port = 59433, sin6_flowinfo =
>>         1136583534, sin6_addr = {__in6_u = {
>>
>>                           __u6_addr8 =
>>         "\360#%\275\016\000\000\000\360?\300\255\177\000",
>>         __u6_addr16 = {9200, 48421, 14, 0, 52976, 49287, 32685, 0},
>>         __u6_addr32 = {3173327856, 14, 3230125808, 32685}}},
>>
>>                       sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}
>>
>>                 newreq = <optimized out>
>>
>>                 msg_buf = <optimized out>
>>
>>         #3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized
>>         out>) at proto_wss.c:428
>>
>>                 size = <optimized out>
>>
>>                 __FUNCTION__ = "wss_read_req"
>>
>>         #4  0x000000000059c374 in handle_io (fm=<optimized out>,
>>         idx=idx at entry=0, event_type=event_type at entry=1) at
>>         net/net_tcp_proc.c:205
>>
>>                 ret = 0
>>
>>                 n = <optimized out>
>>
>>                 con = 0x7fadbeb3f718
>>
>>                 s = 57
>>
>>                 rw = <optimized out>
>>
>>                 resp = <optimized out>
>>
>>                 response = {140384203757736, 1}
>>
>>                 __FUNCTION__ = "handle_io"
>>
>>         #5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized
>>         out>, t=<optimized out>, repeat=<optimized out>) at
>>         net/../io_wait_loop.h:221
>>
>>                 ret = <optimized out>
>>
>>                 e = <optimized out>
>>
>>                 n = 1
>>
>>                 r = 0
>>
>>         #6  tcp_worker_proc (unix_sock=<optimized out>) at
>>         net/net_tcp_proc.c:312
>>
>>                 __FUNCTION__ = "tcp_worker_proc"
>>
>>         #7  0x00000000005a7fc9 in tcp_start_processes
>>         (chd_rank=chd_rank at entry=0x8417e0 <chd_rank.11018>,
>>         startup_done=startup_done at entry=0x7fadbe978738) at
>>         net/net_tcp.c:1761
>>
>>                 r = 5
>>
>>                 reader_fd = {50, 52}
>>
>>                 pid = 0
>>
>>                 load_p = 0x7fadbe979398
>>
>>                 __FUNCTION__ = "tcp_start_processes"
>>
>>         #8  0x0000000000419ef5 in main_loop () at main.c:677
>>
>>                 startup_done = 0x7fadbe978738
>>
>>                 chd_rank = 18
>>
>>         #9  main (argc=<optimized out>, argv=<optimized out>) at
>>         main.c:1249
>>
>>                 cfg_stream = <optimized out>
>>
>>                 c = <optimized out>
>>
>>                 r = <optimized out>
>>
>>                 tmp = 0x7ffc1787bf65 ""
>>
>>                 tmp_len = <optimized out>
>>
>>                 port = <optimized out>
>>
>>                 proto = <optimized out>
>>
>>                 protos_no = <optimized out>
>>
>>                 options = 0x5dae60
>>         "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"
>>
>>         ---Type <return> to continue, or q <return> to quit---
>>
>>                 ret = -1
>>
>>                 seed = 995413301
>>
>>                 rfd = <optimized out>
>>
>>                 __FUNCTION__ = "main"
>>
>>
>>
>>         currently using version: opensips 2.3.0-dev (x86_64/linux)
>>
>>         flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
>>         PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>>         ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
>>         MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>>         poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>
>>         git revision: 92434ef
>>
>>         main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>>
>>
>>         On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu
>>         <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>             Hi Tito,
>>
>>             If opensips crashes, were you able to extract a backtrace
>>             from the core file(s) ?
>>
>>             Thanks and regards,
>>
>>             Bogdan-Andrei Iancu
>>             OpenSIPS Founder and Developer
>>             http://www.opensips-solutions.com
>>
>>             On 27.06.2016 21:20, Tito Cumpen wrote:
>>>             Group,
>>>
>>>
>>>             I am experiencing strange behavior when configuring sip
>>>             tls on port 443. At time opensips crashes or stops
>>>             accepting new connections. Here are the tcp configs I am
>>>             using:
>>>
>>>             #disable_tcp=no
>>>
>>>             tcp_connection_lifetime=3600
>>>
>>>             tcp_connect_timeout=3
>>>
>>>             tcp_keepidle = 30
>>>
>>>             tcp_keepinterval = 5
>>>
>>>             tcp_keepalive = 1
>>>
>>>             tcp_keepcount = 5
>>>
>>>             tcp_max_msg_time = 8
>>>
>>>             tcp_children=10
>>>
>>>
>>>             Any idea what would case this? I am assuming there are
>>>             probes out in the internet that eventually make opensips
>>>             crash?
>>>
>>>             Thanks,
>>>             Tito
>>>
>>>
>>>             _______________________________________________
>>>             Users mailing list
>>>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
>

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From alain.bieuzent at free.fr  Tue Jul  5 14:33:25 2016
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 05 Jul 2016 14:33:25 +0200
Subject: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer
 Probing problem
In-Reply-To: <577B7C07.5090501@opensips.org>
References: <C189FB35-6326-469B-9BF4-7EC3E8C1D53D@free.fr>
	<57764040.7050403@opensips.org>
	<11A9E633-7B12-4A6D-9E5F-3A5819FEAEAC@free.fr>
	<577A428A.6020505@opensips.org>
	<9BD4553E-0058-4066-B2CE-C936841839BE@free.fr>
	<577A980C.9010706@opensips.org>
	<DC35B9D4-4F9F-482C-8EE1-51297E7FE4F0@free.fr>
	<577B7C07.5090501@opensips.org>
Message-ID: <14E9EBB7-7B1C-4012-AB58-A21010198CA5@free.fr>

Hi Bogdan, 

 

Yes i found also some strange issue with the first patch.

The second is better.

 

I will do some other tests on my test server, please tell me when it?s ok for you.

 

Regards,

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : mardi 5 juillet 2016 11:21
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Thanks for your testing and feedback. We found some issues with the patch I sent you yesterday, so please consider for usage this second one (revert the first one please).

Also, I will push this fix on opensips trunk, to get it tested and to be sure there are no side effects.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 05.07.2016 10:24, Alain Bieuzent wrote:

Ho bogdan,

 

Tested your patch this morning, it works.

 

Many thanks.

 

Regards

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : lundi 4 juillet 2016 19:08
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Could you please test the following patch.

Thank and regards,


Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 04.07.2016 15:34, Alain Bieuzent wrote:

Hi Bogdan,

 

Yes i compiled form source.

 

Regards

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : lundi 4 juillet 2016 13:03
? : Alain Bieuzent <alain.bieuzent at free.fr>, "users at lists.opensips.org" <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hi Alain,

Thanks for the trace, I managed to find out the problem. I can provide you a fix to test - do you have OpenSIPS compiled from sources ? or ?

Regards,



Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01.07.2016 16:56, Alain Bieuzent wrote:

Hi Bogdan,

 

As i see in trace (see pcap attach) the first sip OTIONS is send with the correct source IP (..........) to my 2 gateway (see frame 3 and 4) and later there is 2 other sip OPTIONS sent with my private IP (............) (see frame 10 and 11). But all the packet are sending from the same interface (00:50:56:9a:78:93).

 

My network configuration :

 

[....] 

 

 

######################################################################

## Connectivity

######################################################################

[....] 

 

 

Thanks.

 

 

De : Bogdan-Andrei Iancu <bogdan at opensips.org>
Date : vendredi 1 juillet 2016 12:04
? : OpenSIPS users mailling list <users at lists.opensips.org>, Alain Bieuzent <alain.bieuzent at free.fr>
Objet : Re: [OpenSIPS-Users] Migrating from 1.11 to 2.1 load balancer Probing problem

 

Hello Alain,

You mean when comes to sent to ping a single destination, there are actually 2 OPTIONS sent out (on different interfaces) or it alternates (one ping is sent via eth0, the next one via eth1) ?

if you have a pcap to show the problem, please send it to me off-list.

Best regards,




Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 30.06.2016 12:49, Alain Bieuzent wrote:

Hi  all,

 

I?m currently trying to migrate my opensips V1.11.5 to  V2.1.3.

 

My server has 3 interfaces:

eth0 with public IP

eth0:0 with public IP provide via a keepalived script

eth1 with a private IP

 

Opensips listen on IP of eth0:0 and on eth1

Mhomed=no

 

 

Since i migrate to V2.1.3, load balancer probing is UP and down. After checking traffic, i find for each gateway used by load balancer table, there is 2 SIP OPTIONS sent.

-          First is sent with source IP of eth0:0

-          Second is sent with source IP of eth1

 

Any idea ?

 

Thanls







_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
 

 

 

 

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From siju.irs at gmail.com  Tue Jul  5 14:33:25 2016
From: siju.irs at gmail.com (Siju Nair)
Date: Tue, 5 Jul 2016 18:03:25 +0530
Subject: [OpenSIPS-Users] Working features in opensips
Message-ID: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>

Hi Team,
Just wanted to know if we can do below features in opensips with out using
any media servers...

call forwarding (not conditional forwarding) through phone
find me follow me
VM



Thanks,
Siju Nair
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From tito at xsvoce.com  Tue Jul  5 14:02:17 2016
From: tito at xsvoce.com (Tito Cumpen)
Date: Tue, 5 Jul 2016 08:02:17 -0400
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <577BA0C8.8000003@opensips.org>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
	<CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
	<577A325B.1070809@opensips.org>
	<CANZPVB4MMu5N=OuTEVg6vqTksOmcKVRmx4nurei3Fgs9MuW_bg@mail.gmail.com>
	<577BA0C8.8000003@opensips.org>
Message-ID: <CANZPVB7L4Sh_WiYava3XBLdRZm6jX7kOicLaSYGZUdNtP+3KZw@mail.gmail.com>

Bogdan,


Apologies for not providing a full bt. This is a development server with
very little load.

Reading symbols from /usr/sbin/opensips...done.

[New LWP 2685]

[Thread debugging using libthread_db enabled]

Using host libthread_db library "/lib64/libthread_db.so.1".

Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips
-g opensips -M 1024 -f /etc'.

Program terminated with signal 6, Aborted.

#0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at
../nptl/sysdeps/unix/sysv/linux/raise.c:56

56   return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig);

(gdb) bt full

#0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at
../nptl/sysdeps/unix/sysv/linux/raise.c:56

        resultvar = 0

        pid = 2685

        selftid = 2685

#1  0x00007fc95ab1ece8 in __GI_abort () at abort.c:90

        save_stage = 2

        act = {__sigaction_handler = {sa_handler = 0x7fc90eb056c0
<wss_current_req>, sa_sigaction = 0x7fc90eb056c0 <wss_current_req>},
sa_mask = {__val = {140501681180296, 140501713793640,

              140502794949216, 140501683341128, 140501670335743,
140501683341128, 16668947009245032192, 58, 140501683442448,
140501713793689, 65534, 125184, 42698496, 125008, 0, 140502790453088}},

          sa_flags = 1, sa_restorer = 0x0}

        sigs = {__val = {32, 0 <repeats 15 times>}}

#2  0x000000000050d48c in qm_free (qm=<optimized out>,
p=p at entry=0x7fc91abdb298,
file=file at entry=0x7fc90e900108 "../proto_ws/ws_common.h",

    func=func at entry=0x7fc90e903d89 <__FUNCTION__.22006> "ws_process",
line=line at entry=560) at mem/q_malloc.c:469

        f = <optimized out>

        prev = <optimized out>

        next = <optimized out>

        size = <optimized out>

        __FUNCTION__ = "qm_free"

#3  0x00007fc90e8fcd6c in ws_process (con=<optimized out>) at
../proto_ws/ws_common.h:560

        size = 0

        req = <optimized out>

        ret_code = WS_ERR_NONE

        bk = <optimized out>

        msg_len = <optimized out>

        local_rcv = {src_ip = {af = 2, len = 4, u = {addrl = {3870954855,
0}, addr32 = {3870954855, 0, 0, 0}, addr16 = {5479, 59066, 0, 0, 0, 0, 0,
0},

              addr = "g\025\272\346", '\000' <repeats 11 times>}}, dst_ip =
{af = 2, len = 4, u = {addrl = {1252388288, 0}, addr32 = {1252388288, 0, 0,
0}, addr16 = {60864, 19109, 0, 0, 0, 0, 0, 0},

              addr = "\300\355\245J", '\000' <repeats 11 times>}}, src_port
= 57327, dst_port = 10443, proto = 6, proto_reserved1 = 702,
proto_reserved2 = 0, src_su = {s = {sa_family = 2,

              sa_data =
"\337\357g\025\272\346\000\000\000\000\000\000\000"}, sin = {sin_family =
2, sin_port = 61407, sin_addr = {s_addr = 3870954855}, sin_zero =
"\000\000\000\000\000\000\000"},

            sin6 = {sin6_family = 2, sin6_port = 61407, sin6_flowinfo =
3870954855, sin6_addr = {__in6_u = {__u6_addr8 = '\000' <repeats 15 times>,
__u6_addr16 = {0, 0, 0, 0, 0, 0, 0, 0}, __u6_addr32 = {

                    0, 0, 0, 0}}}, sin6_scope_id = 1024}}, bind_address =
0x7fc91ab99738}

        newreq = <optimized out>

        msg_buf = <optimized out>

#4  wss_read_req (con=<optimized out>, bytes_read=<optimized out>) at
proto_wss.c:428

        size = <optimized out>

        __FUNCTION__ = "wss_read_req"

#5  0x000000000059049c in handle_io (fm=<optimized out>, idx=idx at entry=4,
event_type=event_type at entry=1) at net/net_tcp_proc.c:205

        ret = 0

        n = <optimized out>

        con = 0x7fc918ee9148

        s = 58

        rw = <optimized out>

        resp = <optimized out>

        response = {140501683441992, 1}

        __FUNCTION__ = "handle_io"

#6  0x0000000000591a44 in io_wait_loop_epoll (h=<optimized out>,
t=<optimized out>, repeat=<optimized out>) at net/../io_wait_loop.h:221

        ret = <optimized out>

        e = <optimized out>

        n = 1

        r = 4

#7  tcp_worker_proc (unix_sock=<optimized out>) at net/net_tcp_proc.c:312

        __FUNCTION__ = "tcp_worker_proc"

#8  0x000000000059bf09 in tcp_start_processes (chd_rank=chd_rank at entry=0x834800
<chd_rank.11049>, startup_done=startup_done at entry=0x7fc918ce2c60) at
net/net_tcp.c:1761

        r = 4

        reader_fd = {48, 50}

        pid = 0

        load_p = 0x7fc918ce4390

        __FUNCTION__ = "tcp_start_processes"

#9  0x0000000000419417 in main_loop () at main.c:677

        startup_done = 0x7fc918ce2c60

        chd_rank = 17

#10 main (argc=<optimized out>, argv=<optimized out>) at main.c:1249

        cfg_stream = <optimized out>

---Type <return> to continue, or q <return> to quit---

        c = <optimized out>

        r = <optimized out>

        tmp = 0x7ffe75313f65 ""

        tmp_len = <optimized out>

        port = <optimized out>

        proto = <optimized out>

        protos_no = <optimized out>

        options = 0x5ce330 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"

        ret = -1

        seed = 4282461619

        rfd = <optimized out>

        __FUNCTION__ = "main"

On Tue, Jul 5, 2016 at 7:58 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Tito,
>
> Could you print the whole backtrace ?
>
> Thanks,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.07.2016 14:37, Tito Cumpen wrote:
>
> Bogdan,
>
> Here is the backtrace with those flags enabled
>
> gdb /usr/sbin/opensips core.2685
>
> GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-80.el7
>
> Copyright (C) 2013 Free Software Foundation, Inc.
>
> License GPLv3+: GNU GPL version 3 or later <
> http://gnu.org/licenses/gpl.html>
>
> This is free software: you are free to change and redistribute it.
>
> There is NO WARRANTY, to the extent permitted by law.  Type "show copying"
>
> and "show warranty" for details.
>
> This GDB was configured as "x86_64-redhat-linux-gnu".
>
> For bug reporting instructions, please see:
>
> <http://www.gnu.org/software/gdb/bugs/>...
>
> Reading symbols from /usr/sbin/opensips...done.
>
> [New LWP 2685]
>
> [Thread debugging using libthread_db enabled]
>
> Using host libthread_db library "/lib64/libthread_db.so.1".
>
> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u opensips
> -g opensips -M 1024 -f /etc'.
>
> Program terminated with signal 6, Aborted.
>
> #0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at
> ../nptl/sysdeps/unix/sysv/linux/raise.c:56
>
> 56   return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig);
>
> (gdb)
>
>
> Let me know if you need anything else
>
>
>
> On Mon, Jul 4, 2016 at 5:54 AM, Bogdan-Andrei Iancu <
> <bogdan at opensips.org>bogdan at opensips.org> wrote:
>
>> Thank you Tito,
>>
>> It looks like a crash in the memory manager, while trying to allocate a
>> new structure. Do debug such problems there is no other way than enabling
>> debugging for the memory manager (QM_MALLOC + DBG_MALLOC flags) - is this a
>> production system with considerable load on it ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 01.07.2016 18:44, Tito Cumpen wrote:
>>
>> Bogdan,
>>
>>
>> Correction I was using :
>>
>> version: opensips 2.2.0 (x86_64/linux)
>>
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>
>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>
>> git revision: d835721
>>
>> main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5
>>
>>
>>
>>
>>
>> On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen < <tito at xsvoce.com>
>> tito at xsvoce.com> wrote:
>>
>>> Bogdan,
>>>
>>> Here is the backtrace:
>>>
>>>
>>> Reading symbols from /usr/sbin/opensips...done.
>>>
>>> [New LWP 2648]
>>>
>>> [Thread debugging using libthread_db enabled]
>>>
>>> Using host libthread_db library "/lib64/libthread_db.so.1".
>>>
>>> Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u
>>> opensips -g opensips -M 1024 -f /etc'.
>>>
>>> Program terminated with signal 11, Segmentation fault.
>>>
>>> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>>> qm=0x7fadc0809010) at mem/f_malloc.c:187
>>>
>>> 187 *pf=n->u.nxt_free;
>>>
>>> Missing separate debuginfos, use: debuginfo-install
>>> cyrus-sasl-lib-2.1.26-20.el7_2.x86_64 glibc-2.17-106.el7_2.4.x86_64
>>> gmp-6.0.0-12.el7_1.x86_64 gnutls-3.3.8-14.el7_2.x86_64
>>> keyutils-libs-1.5.8-3.el7.x86_64 krb5-libs-1.13.2-12.el7_2.x86_64
>>> libcom_err-1.42.9-7.el7.x86_64 libcurl-7.29.0-25.el7.centos.x86_64
>>> libffi-3.0.13-16.el7.x86_64 libgcc-4.8.5-4.el7.x86_64
>>> libgcrypt-1.5.3-12.el7_1.1.x86_64 libgpg-error-1.12-3.el7.x86_64
>>> libidn-1.28-4.el7.x86_64 libmicrohttpd-0.9.33-2.el7.x86_64
>>> librabbitmq-0.5.2-1.el7.x86_64 libselinux-2.2.2-6.el7.x86_64
>>> libssh2-1.4.3-10.el7_2.1.x86_64 libstdc++-4.8.5-4.el7.x86_64
>>> libtasn1-3.8-2.el7.x86_64 nettle-2.7.1-4.el7.x86_64
>>> nspr-4.11.0-1.el7_2.x86_64 nss-3.21.0-9.el7_2.x86_64
>>> nss-softokn-freebl-3.16.2.3-14.2.el7_2.x86_64
>>> nss-util-3.21.0-2.2.el7_2.x86_64 openldap-2.4.40-9.el7_2.x86_64
>>> openssl-libs-1.0.1e-51.el7_2.4.x86_64 p11-kit-0.20.7-3.el7.x86_64
>>> pcre-8.32-15.el7.x86_64 trousers-0.3.13-1.el7.x86_64
>>> xz-libs-5.1.2-12alpha.el7.x86_64 zlib-1.2.7-15.el7.x86_64
>>>
>>> (gdb) bt full
>>>
>>> #0  0x0000000000512782 in fm_remove_free (n=0x7fadc0898210,
>>> qm=0x7fadc0809010) at mem/f_malloc.c:187
>>>
>>>         pf = 0x0
>>>
>>>         hash = 2051
>>>
>>> #1  fm_malloc (qm=0x7fadc0809010, size=size at entry=65608) at
>>> mem/f_malloc.c:415
>>>
>>>         frag = 0x7fadc0898210
>>>
>>>         n = <optimized out>
>>>
>>>         hash = 2051
>>>
>>>         __FUNCTION__ = "fm_malloc"
>>>
>>> #2  0x00007fadb4615091 in ws_process (con=0x7fadbeb3f718) at
>>> ../proto_ws/ws_common.h:576
>>>
>>>         size = <optimized out>
>>>
>>>         req = <optimized out>
>>>
>>>         ret_code = WS_ERR_NONE
>>>
>>>         bk = <optimized out>
>>>
>>>         msg_len = <optimized out>
>>>
>>>         local_rcv = {src_ip = {af = 3082146304, len = 32685, u = {addrl
>>> = {34359801655, 140720703251544}, addr32 = {63287, 8, 394765400, 32764},
>>> addr16 = {63287, 0, 8, 0, 42072, 6023, 32764, 0},
>>>
>>>               addr =
>>> "7\367\000\000\b\000\000\000X\244\207\027\374\177\000"}}, dst_ip = {af =
>>> 3080002780, len = 32685, u = {addrl = {4, 63119970000}, addr32 = {4, 0,
>>> 2990427856, 14}, addr16 = {4, 0, 0,
>>>
>>>                 0, 20176, 45630, 14, 0}, addr =
>>> "\004\000\000\000\000\000\000\000\320N>\262\016\000\000"}}, src_port =
>>> 52976, dst_port = 49287, proto = 32685, proto_reserved1 = -1214962633,
>>>
>>>           proto_reserved2 = 32685, src_su = {s = {sa_family = 2, sa_data
>>> = ")\350n\343\276C\360#%\275\016\000\000"}, sin = {sin_family = 2, sin_port
>>> = 59433, sin_addr = {s_addr = 1136583534},
>>>
>>>               sin_zero = "\360#%\275\016\000\000"}, sin6 = {sin6_family
>>> = 2, sin6_port = 59433, sin6_flowinfo = 1136583534, sin6_addr = {__in6_u = {
>>>
>>>                   __u6_addr8 =
>>> "\360#%\275\016\000\000\000\360?\300\255\177\000", __u6_addr16 = {9200,
>>> 48421, 14, 0, 52976, 49287, 32685, 0}, __u6_addr32 = {3173327856, 14,
>>> 3230125808, 32685}}},
>>>
>>>               sin6_scope_id = 1}}, bind_address = 0x7fadc0882ea8}
>>>
>>>         newreq = <optimized out>
>>>
>>>         msg_buf = <optimized out>
>>>
>>> #3  wss_read_req (con=0x7fadbeb3f718, bytes_read=<optimized out>) at
>>> proto_wss.c:428
>>>
>>>         size = <optimized out>
>>>
>>>         __FUNCTION__ = "wss_read_req"
>>>
>>> #4  0x000000000059c374 in handle_io (fm=<optimized out>, idx=idx at entry=0,
>>> event_type=event_type at entry=1) at net/net_tcp_proc.c:205
>>>
>>>         ret = 0
>>>
>>>         n = <optimized out>
>>>
>>>         con = 0x7fadbeb3f718
>>>
>>>         s = 57
>>>
>>>         rw = <optimized out>
>>>
>>>         resp = <optimized out>
>>>
>>>         response = {140384203757736, 1}
>>>
>>>         __FUNCTION__ = "handle_io"
>>>
>>> #5  0x000000000059d914 in io_wait_loop_epoll (h=<optimized out>,
>>> t=<optimized out>, repeat=<optimized out>) at net/../io_wait_loop.h:221
>>>
>>>         ret = <optimized out>
>>>
>>>         e = <optimized out>
>>>
>>>         n = 1
>>>
>>>         r = 0
>>>
>>> #6  tcp_worker_proc (unix_sock=<optimized out>) at net/net_tcp_proc.c:312
>>>
>>>         __FUNCTION__ = "tcp_worker_proc"
>>>
>>> #7  0x00000000005a7fc9 in tcp_start_processes (chd_rank=chd_rank at entry=0x8417e0
>>> <chd_rank.11018>, startup_done=startup_done at entry=0x7fadbe978738) at
>>> net/net_tcp.c:1761
>>>
>>>         r = 5
>>>
>>>         reader_fd = {50, 52}
>>>
>>>         pid = 0
>>>
>>>         load_p = 0x7fadbe979398
>>>
>>>         __FUNCTION__ = "tcp_start_processes"
>>>
>>> #8  0x0000000000419ef5 in main_loop () at main.c:677
>>>
>>>         startup_done = 0x7fadbe978738
>>>
>>>         chd_rank = 18
>>>
>>> #9  main (argc=<optimized out>, argv=<optimized out>) at main.c:1249
>>>
>>>         cfg_stream = <optimized out>
>>>
>>>         c = <optimized out>
>>>
>>>         r = <optimized out>
>>>
>>>         tmp = 0x7ffc1787bf65 ""
>>>
>>>         tmp_len = <optimized out>
>>>
>>>         port = <optimized out>
>>>
>>>         proto = <optimized out>
>>>
>>>         protos_no = <optimized out>
>>>
>>>         options = 0x5dae60 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:"
>>>
>>> ---Type <return> to continue, or q <return> to quit---
>>>
>>>         ret = -1
>>>
>>>         seed = 995413301
>>>
>>>         rfd = <optimized out>
>>>
>>>         __FUNCTION__ = "main"
>>>
>>>
>>>
>>> currently using version: opensips 2.3.0-dev (x86_64/linux)
>>>
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>> QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>
>>> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>
>>> git revision: 92434ef
>>>
>>> main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>>>
>>> On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu <
>>> <bogdan at opensips.org>bogdan at opensips.org> wrote:
>>>
>>>> Hi Tito,
>>>>
>>>> If opensips crashes, were you able to extract a backtrace from the core
>>>> file(s) ?
>>>>
>>>> Thanks and regards,
>>>>
>>>> Bogdan-Andrei Iancu
>>>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>>>
>>>> On 27.06.2016 21:20, Tito Cumpen wrote:
>>>>
>>>> Group,
>>>>
>>>>
>>>> I am experiencing strange behavior when configuring sip tls on port
>>>> 443. At time opensips crashes or stops accepting new connections. Here are
>>>> the tcp configs I am using:
>>>>
>>>> #disable_tcp=no
>>>>
>>>> tcp_connection_lifetime=3600
>>>>
>>>> tcp_connect_timeout=3
>>>>
>>>> tcp_keepidle = 30
>>>>
>>>> tcp_keepinterval = 5
>>>>
>>>> tcp_keepalive = 1
>>>>
>>>> tcp_keepcount = 5
>>>>
>>>> tcp_max_msg_time = 8
>>>> tcp_children=10
>>>>
>>>>
>>>> Any idea what would case this? I am assuming there are probes out in
>>>> the internet that eventually make opensips crash?
>>>>
>>>> Thanks,
>>>> Tito
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>>
>>>
>>
>>
>
>
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From bogdan at opensips.org  Tue Jul  5 15:43:01 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 5 Jul 2016 16:43:01 +0300
Subject: [OpenSIPS-Users] running sip tls on 443
In-Reply-To: <CANZPVB7L4Sh_WiYava3XBLdRZm6jX7kOicLaSYGZUdNtP+3KZw@mail.gmail.com>
References: <CANZPVB53DZ=hfWfyui0_yA5Zj51PodfmG6VqCeXw8PW0Fo2VoA@mail.gmail.com>
	<57726D11.5010904@opensips.org>
	<CANZPVB6wkde24GUS0igYywnM-ixEfT0kAENMrKFn_PdLvvp=pQ@mail.gmail.com>
	<CANZPVB6Mvrn0qqYRJNbtJQjYw4pEqPStmcMb6EBxr-EXLJo3Rw@mail.gmail.com>
	<577A325B.1070809@opensips.org>
	<CANZPVB4MMu5N=OuTEVg6vqTksOmcKVRmx4nurei3Fgs9MuW_bg@mail.gmail.com>
	<577BA0C8.8000003@opensips.org>
	<CANZPVB7L4Sh_WiYava3XBLdRZm6jX7kOicLaSYGZUdNtP+3KZw@mail.gmail.com>
Message-ID: <577BB965.4030104@opensips.org>

Many thanks Tito,

Now we know we deal with a pkg_free over an invalid pointer (which does 
not point in the pkg mem pool).

Unfortunately there are a lot of variables not visible due optimization 
- could you (and I promise this is the last time to ask this :D) to 
recompile everything with "mode=debug make all" (add the "mode=debug" 
before make) - this will disable compiling optimizations and all 
variables will be visible in GDB.

After that, a new bt will be more than welcome.

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 15:02, Tito Cumpen wrote:
> Bogdan,
>
>
> Apologies for not providing a full bt. This is a development server 
> with very little load.
>
> Reading symbols from /usr/sbin/opensips...done.
>
> [New LWP 2685]
>
> [Thread debugging using libthread_db enabled]
>
> [...]
>
> On Tue, Jul 5, 2016 at 7:58 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Tito,
>
>     Could you print the whole backtrace ?
>
>     Thanks,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 05.07.2016 14:37, Tito Cumpen wrote:
>>     Bogdan,
>>
>>     Here is the backtrace with those flags enabled
>>
>>     gdb /usr/sbin/opensips core.2685
>>
>>     GNU gdb (GDB) Red Hat Enterprise Linux 7.6.1-80.el7
>>
>>     Copyright (C) 2013 Free Software Foundation, Inc.
>>
>>     License GPLv3+: GNU GPL version 3 or later
>>     <http://gnu.org/licenses/gpl.html>
>>
>>     This is free software: you are free to change and redistribute it.
>>
>>     There is NO WARRANTY, to the extent permitted by law.  Type "show
>>     copying"
>>
>>     and "show warranty" for details.
>>
>>     This GDB was configured as "x86_64-redhat-linux-gnu".
>>
>>     For bug reporting instructions, please see:
>>
>>     <http://www.gnu.org/software/gdb/bugs/>...
>>
>>     Reading symbols from /usr/sbin/opensips...done.
>>
>>     [New LWP 2685]
>>
>>     [Thread debugging using libthread_db enabled]
>>
>>     Using host libthread_db library "/lib64/libthread_db.so.1".
>>
>>     Core was generated by `/sbin/opensips -P /var/run/opensips.pid -u
>>     opensips -g opensips -M 1024 -f /etc'.
>>
>>     Program terminated with signal 6, Aborted.
>>
>>     #0  0x00007fc95ab1d5f7 in __GI_raise (sig=sig at entry=6) at
>>     ../nptl/sysdeps/unix/sysv/linux/raise.c:56
>>
>>     56  return INLINE_SYSCALL (tgkill, 3, pid, selftid, sig);
>>
>>     (gdb)
>>
>>
>>     Let me know if you need anything else
>>
>>
>>
>>     On Mon, Jul 4, 2016 at 5:54 AM, Bogdan-Andrei Iancu
>>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>         Thank you Tito,
>>
>>         It looks like a crash in the memory manager, while trying to
>>         allocate a new structure. Do debug such problems there is no
>>         other way than enabling debugging for the memory manager
>>         (QM_MALLOC + DBG_MALLOC flags) - is this a production system
>>         with considerable load on it ?
>>
>>         Regards,
>>
>>         Bogdan-Andrei Iancu
>>         OpenSIPS Founder and Developer
>>         http://www.opensips-solutions.com
>>
>>         On 01.07.2016 18:44, Tito Cumpen wrote:
>>>         Bogdan,
>>>
>>>
>>>         Correction I was using :
>>>
>>>         version: opensips 2.2.0 (x86_64/linux)
>>>
>>>         flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
>>>         PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>
>>>         ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
>>>         MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>
>>>         poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
>>>
>>>         git revision: d835721
>>>
>>>         main.c compiled on 15:24:26 Jun 28 2016 with gcc 4.8.5
>>>
>>>
>>>
>>>
>>>
>>>         On Fri, Jul 1, 2016 at 9:23 AM, Tito Cumpen <tito at xsvoce.com
>>>         <mailto:tito at xsvoce.com>> wrote:
>>>
>>>             Bogdan,
>>>
>>>             Here is the backtrace:
>>>
>>>
>>>             Reading symbols from /usr/sbin/opensips...done.
>>>
>>>             [New LWP 2648]
>>>
>>>             [...]
>>>
>>>
>>>
>>>             currently using version: opensips 2.3.0-dev (x86_64/linux)
>>>
>>>             flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
>>>             PKG_MALLOC, QM_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>>
>>>             ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
>>>             MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
>>>
>>>             poll method support: poll, epoll_lt, epoll_et, sigio_rt,
>>>             select.
>>>
>>>             git revision: 92434ef
>>>
>>>             main.c compiled on 15:05:06 Jun 28 2016 with gcc 4.8.5
>>>
>>>
>>>             On Tue, Jun 28, 2016 at 8:26 AM, Bogdan-Andrei Iancu
>>>             <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>>
>>>                 Hi Tito,
>>>
>>>                 If opensips crashes, were you able to extract a
>>>                 backtrace from the core file(s) ?
>>>
>>>                 Thanks and regards,
>>>
>>>                 Bogdan-Andrei Iancu
>>>                 OpenSIPS Founder and Developer
>>>                 http://www.opensips-solutions.com
>>>
>>>                 On 27.06.2016 21:20, Tito Cumpen wrote:
>>>>                 Group,
>>>>
>>>>
>>>>                 I am experiencing strange behavior when configuring
>>>>                 sip tls on port 443. At time opensips crashes or
>>>>                 stops accepting new connections. Here are the tcp
>>>>                 configs I am using:
>>>>
>>>>                 #disable_tcp=no
>>>>
>>>>                 tcp_connection_lifetime=3600
>>>>
>>>>                 tcp_connect_timeout=3
>>>>
>>>>                 tcp_keepidle = 30
>>>>
>>>>                 tcp_keepinterval = 5
>>>>
>>>>                 tcp_keepalive = 1
>>>>
>>>>                 tcp_keepcount = 5
>>>>
>>>>                 tcp_max_msg_time = 8
>>>>
>>>>                 tcp_children=10
>>>>
>>>>
>>>>                 Any idea what would case this? I am assuming there
>>>>                 are probes out in the internet that eventually make
>>>>                 opensips crash?
>>>>
>>>>                 Thanks,
>>>>                 Tito
>>>>
>>>>
>>>>                 _______________________________________________
>>>>                 Users mailing list
>>>>                 Users at lists.opensips.org
>>>>                 <mailto:Users at lists.opensips.org>
>>>>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>
>>
>>
>
>

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From liviu at opensips.org  Tue Jul  5 18:54:04 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 5 Jul 2016 19:54:04 +0300
Subject: [OpenSIPS-Users] opensips mysql database connectivity
In-Reply-To: <CAArAGKkvYvU=t7TSphJxRUCNj2k1U-w9kVrdafFswqRe_4R7Lw@mail.gmail.com>
References: <CAArAGKnkCoW=at=T_x0PV0OXNndAHaL2dOGr-OJ2fS=8kw8nDQ@mail.gmail.com>
	<CAArAGKkvYvU=t7TSphJxRUCNj2k1U-w9kVrdafFswqRe_4R7Lw@mail.gmail.com>
Message-ID: <577BE62C.9090904@opensips.org>

Hi Owais,

OpenSIPS will create 1 blocking SQL DB connection per worker, per 
different backend, per unique DB URL. Should this connection be lost 
during runtime, it will attempt X retries (usually 3), before returning 
an error for the query. There is no need for more than 1 connection for 
each worker as it will run every query sequentially.

Starting with OpenSIPS 2.1, if you use async MySQL queries, then every 
worker will open, at most, a pool of "db_max_async_connections" [1] 
connections with every backend and unique DB URL, based on how many 
concurrent queries it needs to perform over there. The connections in 
the async pool are always reused, and the pool may grow if there is 
demand for it.

[1] http://www.opensips.org/Documentation/Script-CoreParameters-2-3#toc39

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 04.07.2016 08:03, Owais Ahmad wrote:
> Does OpenSIPS offer DB connection pooling?
>
> On Fri, Jul 1, 2016 at 10:11 PM, Owais Ahmad <millennium.bug at gmail.com 
> <mailto:millennium.bug at gmail.com>> wrote:
>
>     Hello all,
>
>     I
>     ? am using db_mysql module on opensips version 2.1.3
>
>     Is there a way to limit the number of db connections that opensips
>     initializes with?
>     Also, how can we make non-persistent connections mysql connections
>     with opensips??
>
>
>     Regards,
>     Owais
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Rahul.Gupta at ipc.com  Tue Jul  5 19:43:07 2016
From: Rahul.Gupta at ipc.com (Gupta, Rahul)
Date: Tue, 5 Jul 2016 17:43:07 +0000
Subject: [OpenSIPS-Users] Length Branch in Via header
Message-ID: <CY1PR15MB044121940FF2C8A378BA3C75F4390@CY1PR15MB0441.namprd15.prod.outlook.com>

The branch in Via header is getting truncated in REGISTER. From the code seems like the calculation is made using the following


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From Rahul.Gupta at ipc.com  Tue Jul  5 19:57:58 2016
From: Rahul.Gupta at ipc.com (Gupta, Rahul)
Date: Tue, 5 Jul 2016 17:57:58 +0000
Subject: [OpenSIPS-Users] Truncated Branch in Via
Message-ID: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>

We are using opensips 1.11.5 as a stateless proxy and seeing a truncated Branch in Via for my REGISTER message. After some code digging, the MAX length is calculated using the following formula and seems like its truncating the branch after 55 characters. This is causing the REGISTER to fail in our case. Is there a config level solution to this ?

#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ + 1 /*sep*/ + \
                                MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \
                                1 /*extra space, needed by t_calc_branch*/)


truncated from opensips -->     Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
Via from EndPoint    -->                Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ


Thanks
Rahul Gupta


DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail. E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.
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From bogdan at opensips.org  Wed Jul  6 11:38:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 6 Jul 2016 12:38:08 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
Message-ID: <577CD180.5020802@opensips.org>

Hi Rahul,

That define is used to calculate the the max VIA len when OpenSIPS is 
generating its own VIA headers. That max len does not impact the VIA 
headers which were received.

I do not understand exactly (in your example) what's the flow of that 
VIA header. If you want, send me off-list the pcap/ngrep showing the SIP 
package (before and after OpenSIPS) and how it is affected.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 20:57, Gupta, Rahul wrote:
> We are using opensips 1.11.5 as a stateless proxy and seeing a 
> truncated Branch in Via for my REGISTER message. After some code 
> digging, the MAX length is calculated using the following formula and 
> seems like its truncating the branch after 55 characters. This is 
> causing the REGISTER to fail in our case. Is there a config level 
> solution to this ?
> *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ + 1 
> /*sep*/ + \*
> *                                MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ 
> + \*
> *                                1 /*extra space, needed by 
> t_calc_branch*/)*
> truncated from opensips ? Via: SIP/2.0/UDP 
> XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160 
>
> Via from EndPoint ?                   Via: SIP/2.0/UDP 
> XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ 
>
> *Thanks*
> *Rahul Gupta*
> DISCLAIMER: This e-mail may contain information that is confidential, 
> privileged or otherwise protected from disclosure. If you are not an 
> intended recipient of this e-mail, do not duplicate or redistribute it 
> by any means. Please delete it and any attachments and notify the 
> sender that you have received it in error. Unintended recipients are 
> prohibited from taking action on the basis of information in this 
> e-mail. E-mail messages may contain computer viruses or other defects, 
> may not be accurately replicated on other systems, or may be 
> intercepted, deleted or interfered with without the knowledge of the 
> sender or the intended recipient. If you are not comfortable with the 
> risks associated with e-mail messages, you may decide not to use 
> e-mail to communicate with IPC. IPC reserves the right, to the extent 
> and under circumstances permitted by applicable law, to retain, 
> monitor and intercept e-mail messages to and from its systems.
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From k.pabijanskas at gmail.com  Wed Jul  6 18:44:58 2016
From: k.pabijanskas at gmail.com (Karolis Pabijanskas)
Date: Wed, 6 Jul 2016 17:44:58 +0100
Subject: [OpenSIPS-Users] Weird behaviour when replying to an OPTIONS with a
	Cseq of 0
Message-ID: <CANYNSSLkqcKq+BatEwf74aS-uUqibSrwZy7LfPALim4kxAd3KQ@mail.gmail.com>

Hi List,

We seem to be hitting a strange behaviour when we get an OPTIONS ping with
a Cseq of 0. (latest 1.11 branch).

Our routing script contains this at the very beginning to decline OPTIONS
messages:
route {
    if (is_method("OPTIONS")) {
        sl_send_reply("501", "Method not allowed");
        exit;
    };
   ## blah...
}

If we send this OPTIONS request:

2016-07-06 17:12:05 +0100 : CLIENT_IP:5061 -> OPENSIPS_IP:5060

OPTIONS sip:200 at HOSTNAME:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5061
;branch=z9hG4bK-895-1-0 From: sipp <sip:100 at 127.0.0.1>;tag=1 To: <sip:200@
HOSTNAME:5060> Call-ID: 1-895 at 127.0.0.1 CSeq: 0 OPTIONS Contact: sip:100 at 1
27.0.0.1:5061 Max-Forwards: 100 Content-Length: 0

There is no reply from OpenSIPS. Interestingly, siptrace module is also
running and saving captures in Homer. Homer, actually, is getting a copy of
the generated reply:
2016-07-06 17:12:05 +0100 : OPENSIPS_IP:5060 -> CLIENT_IP:5061
SIP/2.0 501 Method not allowed Via: SIP/2.0/UDP 127.0.0.1:5061;received=
*CLIENT_IP*;branch=z9hG4bK-895-1-0 From: sipp <sip:100 at 127.0.0.1>;tag=1 To:
<sip:200 at HOSTNAME:5060>;tag=06a366df8881a48001f15f72f7138d9f.7522 Call-ID: 1
-895 at 127.0.0.1 CSeq: 0 OPTIONS Server: User Agent String Content-Length: 0

But running a tcpdump on the OpenSIPS host reveals that no actual packet is
ever sent to the client. Debug shows:
Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:core:udp_send:
sendto(sock,0x7f7867aee470,324,0,0x7fff77d00090,16): Operation not
permitted(1)
Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:sl:msg_send: udp_send
failed

OpenSIPS is running as root.

Switching Cseq to 1 in that original OPTIONS message works. But in this
particular case we have no control over the Cseq of the host we are being
pinged from, and need to reply. According to the RFC, Cseq should be a
32bit unsigned integer, so 0 should not be an issue.

Any ideas?

Thanks!
Karolis
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From eric at uphreak.com  Wed Jul  6 18:54:32 2016
From: eric at uphreak.com (Eric Tamme)
Date: Wed, 6 Jul 2016 10:54:32 -0600
Subject: [OpenSIPS-Users] Weird behaviour when replying to an OPTIONS
 with a Cseq of 0
In-Reply-To: <CANYNSSLkqcKq+BatEwf74aS-uUqibSrwZy7LfPALim4kxAd3KQ@mail.gmail.com>
References: <CANYNSSLkqcKq+BatEwf74aS-uUqibSrwZy7LfPALim4kxAd3KQ@mail.gmail.com>
Message-ID: <577D37C8.10307@uphreak.com>

obscuring the ip/hostnames in this makes it difficult to troubleshoot - 
please dont do that in the future.

I am not certain the CSeq is the issue.  I believe that the contact is 
broken, aka it contains localhost, so when opensips tries to route it, 
it fails on udp_send: sendto.

Again, this is my guess, but since you have manipulated and obscured the 
complete sip trace it makes it difficult to confirm.

-Eric

On 07/06/2016 10:44 AM, Karolis Pabijanskas wrote:
> Hi List,
>
> We seem to be hitting a strange behaviour when we get an OPTIONS ping 
> with a Cseq of 0. (latest 1.11 branch).
>
> Our routing script contains this at the very beginning to decline 
> OPTIONS messages:
> route {
>     if (is_method("OPTIONS")) {
>         sl_send_reply("501", "Method not allowed");
>         exit;
>     };
>    ## blah...
> }
>
> If we send this OPTIONS request:
>
> 2016-07-06 17:12:05 +0100 : CLIENT_IP:5061 -> OPENSIPS_IP:5060
> OPTIONSsip:200 at HOSTNAME:5060 SIP/2.0 Via: SIP/2.0/UDP 
> 127.0.0.1:5061;branch=z9hG4bK-895-1-0 From: sipp 
> <sip:100 at 127.0.0.1>;tag=1To: <sip:200 at HOSTNAME:5060> Call-ID: 
> 1-895 at 127.0.0.1CSeq: 0 OPTIONSContact: 
> sip:100 at 127.0.0.1:5061Max-Forwards: 100 Content-Length: 0
>
> There is no reply from OpenSIPS. Interestingly, siptrace module is 
> also running and saving captures in Homer. Homer, actually, is getting 
> a copy of the generated reply:
> 2016-07-06 17:12:05 +0100 : OPENSIPS_IP:5060 -> CLIENT_IP:5061
> SIP/2.0 501 Method not allowed Via: SIP/2.0/UDP 
> 127.0.0.1:5061;received=*CLIENT_IP*;branch=z9hG4bK-895-1-0 From: sipp 
> <sip:100 at 127.0.0.1>;tag=1To: 
> <sip:200 at HOSTNAME:5060>;tag=06a366df8881a48001f15f72f7138d9f.7522 
> Call-ID: 1-895 at 127.0.0.1CSeq: 0 OPTIONS Server: User Agent String 
> Content-Length: 0
> But running a tcpdump on the OpenSIPS host reveals that no actual 
> packet is ever sent to the client. Debug shows:
> Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:core:udp_send: 
> sendto(sock,0x7f7867aee470,324,0,0x7fff77d00090,16): Operation not 
> permitted(1)
> Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:sl:msg_send: 
> udp_send failed
>
> OpenSIPS is running as root.
>
> Switching Cseq to 1 in that original OPTIONS message works. But in 
> this particular case we have no control over the Cseq of the host we 
> are being pinged from, and need to reply. According to the RFC, Cseq 
> should be a 32bit unsigned integer, so 0 should not be an issue.
>
> Any ideas?
>
> Thanks!
> Karolis
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From k.pabijanskas at gmail.com  Wed Jul  6 19:08:21 2016
From: k.pabijanskas at gmail.com (Karolis Pabijanskas)
Date: Wed, 6 Jul 2016 18:08:21 +0100
Subject: [OpenSIPS-Users] Weird behaviour when replying to an OPTIONS
 with a Cseq of 0
In-Reply-To: <577D37C8.10307@uphreak.com>
References: <CANYNSSLkqcKq+BatEwf74aS-uUqibSrwZy7LfPALim4kxAd3KQ@mail.gmail.com>
	<577D37C8.10307@uphreak.com>
Message-ID: <CANYNSSLFfjF2ymdaRQtn-pibOcwkPEfRJ45h6sedNXVTZBZOqA@mail.gmail.com>

Thanks. I will see if I can post the actual packet tomorrow.

The only reason the contact header has 127.0.0.1 is because this test
example is generated with sipp and ir is picking that off from [local_ip]
there. Even with an identical packet, if you change the cseq to 1 it will
work as expected. Hence cseq 0 is the suspect as that can reliably be
replicated with it being the only difference.

While these particular packets were generated, we are actually hitting this
in real world where we have someone constantly pinging us starting with
cseq 0. The packet with cseq 0 never gets a reply. All others (1-100, then
they start from 0 again) get a reply. I'll post a copy of that packet
tomorrow too.

Thanks.
On 6 Jul 2016 5:55 p.m., "Eric Tamme" <eric at uphreak.com> wrote:

> obscuring the ip/hostnames in this makes it difficult to troubleshoot -
> please dont do that in the future.
>
> I am not certain the CSeq is the issue.  I believe that the contact is
> broken, aka it contains localhost, so when opensips tries to route it, it
> fails on udp_send: sendto.
>
> Again, this is my guess, but since you have manipulated and obscured the
> complete sip trace it makes it difficult to confirm.
>
> -Eric
>
> On 07/06/2016 10:44 AM, Karolis Pabijanskas wrote:
>
> Hi List,
>
> We seem to be hitting a strange behaviour when we get an OPTIONS ping with
> a Cseq of 0. (latest 1.11 branch).
>
> Our routing script contains this at the very beginning to decline OPTIONS
> messages:
> route {
>     if (is_method("OPTIONS")) {
>         sl_send_reply("501", "Method not allowed");
>         exit;
>     };
>    ## blah...
> }
>
> If we send this OPTIONS request:
>
> 2016-07-06 17:12:05 +0100 : CLIENT_IP:5061 -> OPENSIPS_IP:5060
> OPTIONS sip:200 at HOSTNAME:5060 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5061
> ;branch=z9hG4bK-895-1-0 From: sipp <sip:100 at 127.0.0.1>;tag=1 To: <sip:200@
> HOSTNAME:5060> Call-ID: 1-895 at 127.0.0.1 CSeq: 0 OPTIONS Contact: sip:100 at 1
> 27.0.0.1:5061 Max-Forwards: 100 Content-Length: 0
>
> There is no reply from OpenSIPS. Interestingly, siptrace module is also
> running and saving captures in Homer. Homer, actually, is getting a copy of
> the generated reply:
> 2016-07-06 17:12:05 +0100 : OPENSIPS_IP:5060 -> CLIENT_IP:5061
> SIP/2.0 501 Method not allowed Via: SIP/2.0/UDP 127.0.0.1:5061;received=
> *CLIENT_IP*;branch=z9hG4bK-895-1-0 From: sipp <sip:100 at 127.0.0.1>;tag=1
> To: <sip:200 at HOSTNAME:5060>;tag=06a366df8881a48001f15f72f7138d9f.7522
> Call-ID: 1-895 at 127.0.0.1 CSeq: 0 OPTIONS Server: User Agent String
> Content-Length: 0
> But running a tcpdump on the OpenSIPS host reveals that no actual packet
> is ever sent to the client. Debug shows:
> Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:core:udp_send:
> sendto(sock,0x7f7867aee470,324,0,0x7fff77d00090,16): Operation not
> permitted(1)
> Jul  6 17:40:25 HOSTNAME /sbin/opensips[48357]: ERROR:sl:msg_send:
> udp_send failed
>
> OpenSIPS is running as root.
>
> Switching Cseq to 1 in that original OPTIONS message works. But in this
> particular case we have no control over the Cseq of the host we are being
> pinged from, and need to reply. According to the RFC, Cseq should be a
> 32bit unsigned integer, so 0 should not be an issue.
>
> Any ideas?
>
> Thanks!
> Karolis
>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Agalya_Ramachandran at comcast.com  Wed Jul  6 21:13:48 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 6 Jul 2016 19:13:48 +0000
Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls
In-Reply-To: <daafd55e72df49b8894644c5fd6416df@COPDCEX28.cable.comcast.com>
References: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com>
	<576BB123.7020102@opensips.org>
	<daafd55e72df49b8894644c5fd6416df@COPDCEX28.cable.comcast.com>
Message-ID: <e1287512060e463fa639b7153e5b1de8@COPDCEX28.cable.comcast.com>

Hi Liviu,

On the below topic, I have double checked that call back function is not at called in the case of failure response.
This is confirmed by adding some debug statements in write_func(). I have not made any changes with respect to functionality.
Only in success case I could able to see the call back called.
I could also confirm by seeing the Wireshark traces am getting response body in the case of failure as well.

In our case since call back function is not called, hence no response body is returned.
Can you please check in your end whether call back is called in the case when curl_easy_perform fails.

Regards,
Agalya

From: Ramachandran, Agalya (Contractor)
Sent: Friday, June 24, 2016 5:03 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: RE: [OpenSIPS-Users] CURL library with respect to REST_API calls

Hi Liviu,

I have not changed anything in the write_func(). Also I tried by adding some debug statements.
I observed that only in the case of success write_func() is being called and I get the debug statements in my logs.
In the case of failure, the callback function is never called. This is what I observe.
If you try for failure case, let me know how it worked for you.

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Thursday, June 23, 2016 5:52 AM
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] CURL library with respect to REST_API calls

Hi, Ramachandran!

I read the "CURLOPT_WRITEFUNCTION" section [1] one more time, and it really looks like it should pass the body of the reply to the "write_func" callback we register before sending the HTTP PUT, even if we got an error code (3XX or higher).

Did you change anything in the "write_func()"? Notice how their docs say that if a proper "len" is not returned, the transfer will be aborted. Apart from that, I have no other ideas for now but to try and fetch the body myself on an error HTTP ret code of a CUROPT_PUT operation, see how (or if) it works for me, and give you more feedback.

[1]: https://curl.haxx.se/libcurl/c/CURLOPT_WRITEFUNCTION.html

All the best,

Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 22.06.2016 20:19, Ramachandran, Agalya (Contractor) wrote:
Hi team,

Any one has any clue on the below topic?

Regards,
Agalya

From: Ramachandran, Agalya (Contractor)
Sent: Monday, June 20, 2016 1:45 PM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: CURL library with respect to REST_API calls

Hi team,

I have a question regarding curl library behavior with respect to curl_easy_perform API call.
Here is the snippet of the code that am using in "rest_put" API call in rest_methods.c file

  w_curl_easy_setopt(handle, CURLOPT_WRITEFUNCTION, write_func);
 w_curl_easy_setopt(handle, CURLOPT_WRITEDATA, &res_body);

When curl_easy_perform API call is success, I could able to retrieve the result body from the res_body.
But in the case of API call failure am not getting any details of the message. But getting only the http response code.
Is there a way to get the message details as well in the case where curl_easy_perform API fails?

LM_INFO(" Actual result body is %s\n", res_body.s);
When I print this, in the case of success, am getting a http response message in detail.
But in case of failure, the call back function write_func is not at all called. Why it is so?

Please guide me if there is a way to the message details in case of failure too.

Regards,
Agalya




_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From jeff.pyle at fidelityvoice.com  Thu Jul  7 03:42:06 2016
From: jeff.pyle at fidelityvoice.com (Jeff Pyle)
Date: Wed, 6 Jul 2016 21:42:06 -0400
Subject: [OpenSIPS-Users] return code from perl_exec() versus
	perl_exec_simple()
Message-ID: <CACYJG3J-D_ybc6uu91LVsFXVdJxh6gOV-iYh6U2=xWGaEQ47cA@mail.gmail.com>

Hello,

I'm using 2.1 cloned from git earlier today.  I notice when I use
perl_exec_simple(), the return code is always 1, regardless of the return
code at the end of the perl function.  With perl_exec(), however, the
return code matches the return code in the function.  Is this difference
intentional?


- Jeff
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From siju.irs at gmail.com  Thu Jul  7 07:49:34 2016
From: siju.irs at gmail.com (Siju Nair)
Date: Thu, 7 Jul 2016 11:19:34 +0530
Subject: [OpenSIPS-Users] Fwd: Working features in opensips
In-Reply-To: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
Message-ID: <CAETdsxjRcxPXFq+QV=2uxZ0FdkxUtPZY2G8uCoL381Qco4ypGA@mail.gmail.com>

---------- Forwarded message ----------
From: Siju Nair <siju.irs at gmail.com>
Date: Tue, Jul 5, 2016 at 6:03 PM
Subject: Working features in opensips
To: users at lists.opensips.org


Hi Team,
Just wanted to know if we can do below features in opensips with out using
any media servers...

call forwarding (not conditional forwarding) through phone
find me follow me
VM



Thanks,
Siju Nair
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From bogdan at opensips.org  Thu Jul  7 10:43:21 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jul 2016 11:43:21 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
Message-ID: <577E1629.6070209@opensips.org>

Hi Rahul,

The received VIA hdr (with the .44 IP) is properly preserved when 
forwarding (in the outbound message).

The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and its 
branch value is completely independent from the branch of other VIA hdrs.

Why do they look the same ? The RFC3261 says that for stateless fwd 
(when basically there is no transaction, so no branch value), to avoid 
populating its VIA with ";branch=0" , the proxy may"copy" and use a 
branch value from an older VIA (a received VIA)- keep in mind it does 
not say to copy it entirely or part. So, OpenSIPS copies a 
MAX_BRANCH_PARAM_LEN length string from the previous branch param.

Everything is correct and legal (from RFC perspective).

PS: if you would use t_relay() instead of forward() - doing statefull 
proxy -, you will see that the branch in the VIA added by OpenSIPS will 
be completly different from the value in the previous VIA.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.07.2016 03:42, Gupta, Rahul wrote:
>
> Hi Bogdan,
>
> Here is a Ethernet trace (pcap) file that has a successful and an 
> unsuccessful registration.
>
> Frames 1-8 illustrate the successful case with Frames 2 and 6 show how 
> opensips adds an extra VIA header that has a full VIA;branch
>
> Frame:2
>
> Via: SIP/2.0/UDP 
> 10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
> Via: SIP/2.0/UDP 
> 10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
> Frames 9-16 illustrate the unsuccessful case where Frames 10 and 14 
> show how opensips adds an extra VIA header that has a truncated branch.
>
> Frame 10:
>
> Via: SIP/2.0/UDP 
> 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
> Via: SIP/2.0/UDP 
> 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
> *Thanks*
>
> *Rahul Gupta***
>
> Systems Architect
>
> *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com <mailto:rahul.gupta at ipc.com>
>
> cid:image006.jpg at 01D1940F.3E021840
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Wednesday, July 06, 2016 5:38 AM
> *To:* OpenSIPS users mailling list
> *Cc:* Elliott, Ray; Gupta, Rahul
> *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
> Hi Rahul,
>
> That define is used to calculate the the max VIA len when OpenSIPS is 
> generating its own VIA headers. That max len does not impact the VIA 
> headers which were received.
>
> I do not understand exactly (in your example) what's the flow of that 
> VIA header. If you want, send me off-list the pcap/ngrep showing the 
> SIP package (before and after OpenSIPS) and how it is affected.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 05.07.2016 20:57, Gupta, Rahul wrote:
>
>     We are using opensips 1.11.5 as a stateless proxy and seeing a
>     truncated Branch in Via for my REGISTER message. After some code
>     digging, the MAX length is calculated using the following formula
>     and seems like its truncating the branch after 55 characters. This
>     is causing the REGISTER to fail in our case. Is there a config
>     level solution to this ?
>
>     *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ + 1
>     /*sep*/ + \*
>
>     *MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \*
>
>     *1 /*extra space, needed by t_calc_branch*/)*
>
>     truncated from opensips ?Via: SIP/2.0/UDP
>     XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
>
>
>     Via from EndPoint ?Via: SIP/2.0/UDP
>     XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ
>
>
>     *Thanks*
>
>     *Rahul Gupta*
>
>     DISCLAIMER: This e-mail may contain information that is
>     confidential, privileged or otherwise protected from disclosure.
>     If you are not an intended recipient of this e-mail, do not
>     duplicate or redistribute it by any means. Please delete it and
>     any attachments and notify the sender that you have received it in
>     error. Unintended recipients are prohibited from taking action on
>     the basis of information in this e-mail. E-mail messages may
>     contain computer viruses or other defects, may not be accurately
>     replicated on other systems, or may be intercepted, deleted or
>     interfered with without the knowledge of the sender or the
>     intended recipient. If you are not comfortable with the risks
>     associated with e-mail messages, you may decide not to use e-mail
>     to communicate with IPC. IPC reserves the right, to the extent and
>     under circumstances permitted by applicable law, to retain,
>     monitor and intercept e-mail messages to and from its systems.
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From liviu at opensips.org  Thu Jul  7 10:52:52 2016
From: liviu at opensips.org (Liviu Chircu)
Date: Thu, 7 Jul 2016 11:52:52 +0300
Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls
In-Reply-To: <e1287512060e463fa639b7153e5b1de8@COPDCEX28.cable.comcast.com>
References: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com>
	<576BB123.7020102@opensips.org>
	<daafd55e72df49b8894644c5fd6416df@COPDCEX28.cable.comcast.com>
	<e1287512060e463fa639b7153e5b1de8@COPDCEX28.cable.comcast.com>
Message-ID: <577E1864.3090200@opensips.org>

Although the official documentation specifically states that the default 
libcurl behavior is to return the body even on an error response [1], 
have you tried to explicitly set the CURLOLPT_FAILONERROR option to 0?

[1]: https://curl.haxx.se/libcurl/c/CURLOPT_FAILONERROR.html

Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com

On 06.07.2016 22:13, Ramachandran, Agalya (Contractor) wrote:
Hi Liviu,

On the below topic, I have double checked that call back function is not 
at called in the case of failure response. This is confirmed by adding 
some debug statements in write_func(). I have not made any changes with 
respect to functionality.Only in success case I could able to see the 
call back called.

I could also confirm by seeing the Wireshark traces am getting response 
body in the case of failure as well. In our case since call back 
function is not called, hence no response body is returned. Can you 
please check in your end whether call back is called in the case when 
curl_easy_perform fails.

Regards,
Agalya

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From bogdan at opensips.org  Thu Jul  7 11:44:12 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jul 2016 12:44:12 +0300
Subject: [OpenSIPS-Users] return code from perl_exec() versus
 perl_exec_simple()
In-Reply-To: <CACYJG3J-D_ybc6uu91LVsFXVdJxh6gOV-iYh6U2=xWGaEQ47cA@mail.gmail.com>
References: <CACYJG3J-D_ybc6uu91LVsFXVdJxh6gOV-iYh6U2=xWGaEQ47cA@mail.gmail.com>
Message-ID: <577E246C.9060308@opensips.org>

Hi Jeff,

I think the difference is a mis. Could you please test the attached patch.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.07.2016 04:42, Jeff Pyle wrote:
> Hello,
>
> I'm using 2.1 cloned from git earlier today.  I notice when I use 
> perl_exec_simple(), the return code is always 1, regardless of the 
> return code at the end of the perl function.  With perl_exec(), 
> however, the return code matches the return code in the function.  Is 
> this difference intentional?
>
>
> - Jeff
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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diff --git a/modules/perl/perlfunc.c b/modules/perl/perlfunc.c
index f3b0f2e..68a9509 100644
--- a/modules/perl/perlfunc.c
+++ b/modules/perl/perlfunc.c
@@ -54,18 +54,37 @@ int perl_checkfnc(char *fnc) {
  * Run function without parameters
  */
 
-int perl_exec_simple(char* fnc, char* args[], int flags) {
+int perl_exec_simple(char* fnc, char* args[], int flags)
+{
+	dSP;
+	int ret;
+	int cnt;
 
 	if (perl_checkfnc(fnc)) {
 		LM_DBG("running perl function \"%s\"", fnc);
 
-		call_argv(fnc, flags, args);
-	} else {
-		LM_ERR("unknown function '%s' called.\n", fnc);
-		return -1;
-	}
+		ENTER;
+		SAVETMPS;
+
+		cnt = call_argv(fnc, flags, args);
 
-	return 1;
+		SPAGAIN;
+
+		if (cnt!=1) {
+			LM_ERR("function %s failed to return anything\n",fnc);
+			ret = -1;
+		} else {
+			ret = POPi;
+			PUTBACK;
+		}
+
+		FREETMPS;
+		LEAVE;
+
+		return ret;
+	}
+	LM_ERR("unknown function '%s' called.\n", fnc);
+	return -1;
 }
 
 int perl_exec_simple1(struct sip_msg* _msg, char* fnc, char* str2) {

From bogdan at opensips.org  Thu Jul  7 11:46:09 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jul 2016 12:46:09 +0300
Subject: [OpenSIPS-Users] Working features in opensips
In-Reply-To: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
Message-ID: <577E24E1.1070406@opensips.org>

Hi Siju,

Yes,
Yes,
No (opensips is not doing media).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 15:33, Siju Nair wrote:
> Hi Team,
> Just wanted to know if we can do below features in opensips with out 
> using any media servers...
>
> call forwarding (not conditional forwarding) through phone
> find me follow me
> VM
>
>
>
> Thanks,
> Siju Nair
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From jeff.pyle at fidelityvoice.com  Thu Jul  7 17:29:30 2016
From: jeff.pyle at fidelityvoice.com (Jeff Pyle)
Date: Thu, 7 Jul 2016 11:29:30 -0400
Subject: [OpenSIPS-Users] return code from perl_exec() versus
	perl_exec_simple()
In-Reply-To: <577E246C.9060308@opensips.org>
References: <CACYJG3J-D_ybc6uu91LVsFXVdJxh6gOV-iYh6U2=xWGaEQ47cA@mail.gmail.com>
	<577E246C.9060308@opensips.org>
Message-ID: <CACYJG3KWrXjLiLyqYqnBbe5jt=WMzNNGrk4oHMy5snfVvnNMfQ@mail.gmail.com>

Hi Bogdan,

The patch gives errors.

I have two perl functions:

sub return1 {
return 1;
}

sub returnminus1 {
return -1;
}

In the OpenSIPS script I have:

route {
        perl_exec_simple("return1");
        xlog("L_INFO", "$retcode\n");
        perl_exec_simple("returnminus1");
        xlog("L_INFO", "$retcode\n");

        perl_exec("return1");
        xlog("L_INFO", "$retcode\n");
        perl_exec("returnminus1");
        xlog("L_INFO", "$retcode\n");

        sl_send_reply("600", "Road Closed");
        drop;
        exit;
}

Syslog shows:

 /usr/sbin/opensips[22354]: ERROR:perl:perl_exec_simple: function return1
failed to return anything
 /usr/sbin/opensips[22354]: -1
 /usr/sbin/opensips[22354]: ERROR:perl:perl_exec_simple: function
returnminus1 failed to return anything
 /usr/sbin/opensips[22354]: -1
 /usr/sbin/opensips[22354]: 1
 /usr/sbin/opensips[22354]: -1

perl_exec() still works okay, but perl_exec_simple() gives errors.


- Jeff





On Thu, Jul 7, 2016 at 5:44 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Jeff,
>
> I think the difference is a mis. Could you please test the attached patch.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 07.07.2016 04:42, Jeff Pyle wrote:
>
> Hello,
>
> I'm using 2.1 cloned from git earlier today.  I notice when I use
> perl_exec_simple(), the return code is always 1, regardless of the return
> code at the end of the perl function.  With perl_exec(), however, the
> return code matches the return code in the function.  Is this difference
> intentional?
>
>
> - Jeff
>
>
>
>
>
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From bogdan at opensips.org  Thu Jul  7 17:37:34 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jul 2016 18:37:34 +0300
Subject: [OpenSIPS-Users] return code from perl_exec() versus
 perl_exec_simple()
In-Reply-To: <CACYJG3KWrXjLiLyqYqnBbe5jt=WMzNNGrk4oHMy5snfVvnNMfQ@mail.gmail.com>
References: <CACYJG3J-D_ybc6uu91LVsFXVdJxh6gOV-iYh6U2=xWGaEQ47cA@mail.gmail.com>
	<577E246C.9060308@opensips.org>
	<CACYJG3KWrXjLiLyqYqnBbe5jt=WMzNNGrk4oHMy5snfVvnNMfQ@mail.gmail.com>
Message-ID: <577E773E.4050101@opensips.org>

Jeff,

Can you replace
     LM_ERR("function %s failed to return anything\n",fnc);
with
     LM_ERR("function %s failed to return anything (reports %d) 
\n",fnc,cnt);

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.07.2016 18:29, Jeff Pyle wrote:
> Hi Bogdan,
>
> The patch gives errors.
>
> I have two perl functions:
>
> sub return1 {
> return 1;
> }
>
> sub returnminus1 {
> return -1;
> }
>
> In the OpenSIPS script I have:
>
> route {
>         perl_exec_simple("return1");
>         xlog("L_INFO", "$retcode\n");
>         perl_exec_simple("returnminus1");
>         xlog("L_INFO", "$retcode\n");
>
>         perl_exec("return1");
>         xlog("L_INFO", "$retcode\n");
>         perl_exec("returnminus1");
>         xlog("L_INFO", "$retcode\n");
>
>         sl_send_reply("600", "Road Closed");
>         drop;
>         exit;
> }
>
> Syslog shows:
>
>  /usr/sbin/opensips[22354]: ERROR:perl:perl_exec_simple: function 
> return1 failed to return anything
>  /usr/sbin/opensips[22354]: -1
>  /usr/sbin/opensips[22354]: ERROR:perl:perl_exec_simple: function 
> returnminus1 failed to return anything
>  /usr/sbin/opensips[22354]: -1
>  /usr/sbin/opensips[22354]: 1
>  /usr/sbin/opensips[22354]: -1
>
> perl_exec() still works okay, but perl_exec_simple() gives errors.
>
>
> - Jeff
>
>
>
>
>
> On Thu, Jul 7, 2016 at 5:44 AM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Jeff,
>
>     I think the difference is a mis. Could you please test the
>     attached patch.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 07.07.2016 04:42, Jeff Pyle wrote:
>>     Hello,
>>
>>     I'm using 2.1 cloned from git earlier today.  I notice when I use
>>     perl_exec_simple(), the return code is always 1, regardless of
>>     the return code at the end of the perl function.  With
>>     perl_exec(), however, the return code matches the return code in
>>     the function.  Is this difference intentional?
>>
>>
>>     - Jeff
>>
>>
>>
>
>

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From Agalya_Ramachandran at comcast.com  Thu Jul  7 19:53:34 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 7 Jul 2016 17:53:34 +0000
Subject: [OpenSIPS-Users] CURL library with respect to REST_API calls
In-Reply-To: <577E1864.3090200@opensips.org>
References: <1dc72c09449b413a84275d01bcff3548@COPDCEX28.cable.comcast.com>
	<576BB123.7020102@opensips.org>
	<daafd55e72df49b8894644c5fd6416df@COPDCEX28.cable.comcast.com>
	<e1287512060e463fa639b7153e5b1de8@COPDCEX28.cable.comcast.com>
	<577E1864.3090200@opensips.org>
Message-ID: <3502b5a317c34ae48f2c38c86726b782@COPDCEX28.cable.comcast.com>

Hi Liviu,

I have tried by explicitly changing the option CURLOLPT_FAILONERROR to 0 and tested.
Am getting the http response as 200Ok and I could able to see that call back function is been called and it gives the complete response body.

One more issue observed is, when CURLOLPT_FAILONERROR option is set as 0, right after the response is received, opensips is being crashed.

Where is the path the core dump will be generated? Because am seeing crash onli in logs and if I check the process it is not running. But not seeing any core dump.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Liviu Chircu
Sent: Thursday, July 07, 2016 4:53 AM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] CURL library with respect to REST_API calls

Although the official documentation specifically states that the default libcurl behavior is to return the body even on an error response [1], have you tried to explicitly set the CURLOLPT_FAILONERROR option to 0?

[1]: https://curl.haxx.se/libcurl/c/CURLOPT_FAILONERROR.html


Liviu Chircu

OpenSIPS Developer

http://www.opensips-solutions.com
On 06.07.2016 22:13, Ramachandran, Agalya (Contractor) wrote:
Hi Liviu,
On the below topic, I have double checked that call back function is not at called in the case of failure response. This is confirmed by adding some debug statements in write_func(). I have not made any changes with respect to functionality. Only in success case I could able to see the call back called.
I could also confirm by seeing the Wireshark traces am getting response body in the case of failure as well. In our case since call back function is not called, hence no response body is returned. Can you please check in your end whether call back is called in the case when curl_easy_perform fails.
Regards,
Agalya
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From nocbgtelcom at gmail.com  Fri Jul  8 15:05:52 2016
From: nocbgtelcom at gmail.com (Hristo Donev)
Date: Fri, 8 Jul 2016 16:05:52 +0300
Subject: [OpenSIPS-Users] Number validation
Message-ID: <CAAf1xX-hJYHMP940mNanRHXt-Nob3NyHgn2cBzZA5XzvqLU3Eg@mail.gmail.com>

Hello,

I thinking of performing number validation before I pass the request to the
upstream provider in order to minimize capacity waste because of malformed
numbers. Can this be something I can do in OpenSIPS?
Also, any idea of  good source for this validation for global
(international) numbers?

Best regards,
Hristo
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From eric at uphreak.com  Fri Jul  8 15:08:26 2016
From: eric at uphreak.com (Eric Tamme)
Date: Fri, 8 Jul 2016 07:08:26 -0600
Subject: [OpenSIPS-Users] Number validation
In-Reply-To: <CAAf1xX-hJYHMP940mNanRHXt-Nob3NyHgn2cBzZA5XzvqLU3Eg@mail.gmail.com>
References: <CAAf1xX-hJYHMP940mNanRHXt-Nob3NyHgn2cBzZA5XzvqLU3Eg@mail.gmail.com>
Message-ID: <577FA5CA.4060708@uphreak.com>

yes, https://www.itu.int/rec/T-REC-E.164-201011-I/en


On 07/08/2016 07:05 AM, Hristo Donev wrote:
> Hello,
>
> I thinking of performing number validation before I pass the request 
> to the upstream provider in order to minimize capacity waste because 
> of malformed numbers. Can this be something I can do in OpenSIPS?
> Also, any idea of  good source for this validation for global 
> (international) numbers?
>
> Best regards,
> Hristo
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From rrobson at greenlightcrm.com  Fri Jul  8 15:42:27 2016
From: rrobson at greenlightcrm.com (Richard Robson)
Date: Fri, 8 Jul 2016 14:42:27 +0100
Subject: [OpenSIPS-Users] crash in 2.2
Message-ID: <33960e21-00ee-7286-aa7b-3f6e5530dd91@greenlightcrm.com>

Hi Guys,


I'm testing ratelimiting and when I ramp up to about 4 or 5 cps opensips
crashes. I'm not sure if its the ratelimiting module or opensips itself
crashing dur to the number of calls.

its version 2.2 from the download page dated 16-Jun-2016 18:41 

here is the back trace:

http://pastebin.com/yAwmYyhx

Regards,

-- 
Richard Robson
Greenlight Support
01382 843843
support at greenlightcrm.com



From bogdan at opensips.org  Fri Jul  8 17:15:33 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 8 Jul 2016 18:15:33 +0300
Subject: [OpenSIPS-Users] crash in 2.2
In-Reply-To: <33960e21-00ee-7286-aa7b-3f6e5530dd91@greenlightcrm.com>
References: <33960e21-00ee-7286-aa7b-3f6e5530dd91@greenlightcrm.com>
Message-ID: <577FC395.1030408@opensips.org>

Hi Richard,

The crash indicates the siptrace module. The tarball ia a bit older, 
please try to grab directly the sources from GIT branch 2.2 - this will 
give you the latest sources.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 08.07.2016 16:42, Richard Robson wrote:
> Hi Guys,
>
>
> I'm testing ratelimiting and when I ramp up to about 4 or 5 cps opensips
> crashes. I'm not sure if its the ratelimiting module or opensips itself
> crashing dur to the number of calls.
>
> its version 2.2 from the download page dated 16-Jun-2016 18:41
>
> here is the back trace:
>
> http://pastebin.com/yAwmYyhx
>
> Regards,
>



From Rahul.Gupta at ipc.com  Thu Jul  7 21:01:12 2016
From: Rahul.Gupta at ipc.com (Gupta, Rahul)
Date: Thu, 7 Jul 2016 19:01:12 +0000
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <577E1629.6070209@opensips.org>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
Message-ID: <CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>

Hi Bogdan, so the problem we are facing is the endpoint has a new branch however the difference in the branch value is AFTER the MAX_BRANCH_PARAM_LEN, the highlighted text. Since opensips is copying the partial branch which is same as the previous Via and thus our SIP stack (OCCAS) is sending back 401 Unauthorized for both thinking it's a retransmit.

This is also in line with RFC3261. Shouldn't it copy the entire Via to make sure it's different from previous Message ?

REGISTER 1
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ

REGISTER 2
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915656050004_MTAuMTIuMy4xMQ


Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]



From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, July 07, 2016 4:43 AM
To: Gupta, Rahul
Cc: Elliott, Ray; users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

The received VIA hdr (with the .44 IP) is properly preserved when forwarding (in the outbound message).

The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and its branch value is completely independent from the branch of other VIA hdrs.

Why do they look the same ? The RFC3261 says that for stateless fwd (when basically there is no transaction, so no branch value), to avoid populating its VIA with ";branch=0" , the proxy may "copy" and use a branch value from an older VIA (a received VIA) - keep in mind it does not say to copy it entirely or part. So, OpenSIPS copies a MAX_BRANCH_PARAM_LEN length string from the previous branch param.

Everything is correct and legal (from RFC perspective).

PS: if you would use t_relay() instead of forward() - doing statefull proxy -, you will see that the branch in the VIA added by OpenSIPS will be completly different from the value in the previous VIA.

Best regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 07.07.2016 03:42, Gupta, Rahul wrote:
Hi Bogdan,

Here is a Ethernet trace (pcap) file that has a successful and an unsuccessful registration.

Frames 1-8 illustrate the successful case with Frames 2 and 6 show how opensips adds an extra VIA header that has a full VIA;branch
Frame:2
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
Via: SIP/2.0/UDP 10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8

Frames 9-16 illustrate the unsuccessful case where Frames 10 and 14 show how opensips adds an extra VIA header that has a truncated branch.
Frame 10:
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ

Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]



From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, July 06, 2016 5:38 AM
To: OpenSIPS users mailling list
Cc: Elliott, Ray; Gupta, Rahul
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

That define is used to calculate the the max VIA len when OpenSIPS is generating its own VIA headers. That max len does not impact the VIA headers which were received.

I do not understand exactly (in your example) what's the flow of that VIA header. If you want, send me off-list the pcap/ngrep showing the SIP package (before and after OpenSIPS) and how it is affected.

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 05.07.2016 20:57, Gupta, Rahul wrote:
We are using opensips 1.11.5 as a stateless proxy and seeing a truncated Branch in Via for my REGISTER message. After some code digging, the MAX length is calculated using the following formula and seems like its truncating the branch after 55 characters. This is causing the REGISTER to fail in our case. Is there a config level solution to this ?

#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ + 1 /*sep*/ + \
                                MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \
                                1 /*extra space, needed by t_calc_branch*/)


truncated from opensips -->       Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
Via from EndPoint    -->                   Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ


Thanks
Rahul Gupta


DISCLAIMER: This e-mail may contain information that is confidential, privileged or otherwise protected from disclosure. If you are not an intended recipient of this e-mail, do not duplicate or redistribute it by any means. Please delete it and any attachments and notify the sender that you have received it in error. Unintended recipients are prohibited from taking action on the basis of information in this e-mail. E-mail messages may contain computer viruses or other defects, may not be accurately replicated on other systems, or may be intercepted, deleted or interfered with without the knowledge of the sender or the intended recipient. If you are not comfortable with the risks associated with e-mail messages, you may decide not to use e-mail to communicate with IPC. IPC reserves the right, to the extent and under circumstances permitted by applicable law, to retain, monitor and intercept e-mail messages to and from its systems.





_______________________________________________

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http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Fri Jul  8 16:30:47 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 8 Jul 2016 17:30:47 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
Message-ID: <577FB917.6020400@opensips.org>

Hi Rahul,

Indeed, that is a good pointhere, thanks for bring it up. Let me 
investigate the code a bit and I will update you.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.07.2016 22:01, Gupta, Rahul wrote:
>
> Hi Bogdan, so the problem we are facing is the endpoint has a new 
> branch however the difference in the branch value is *AFTER* the 
> MAX_BRANCH_PARAM_LEN, the highlighted text.Since opensips is copying 
> the partial branch which is same as the previous Via and thus our SIP 
> stack (OCCAS) is sending back 401 Unauthorized for both thinking it?s 
> a retransmit.
>
> This is also in line with RFC3261. Shouldn?t it copy the entire Via to 
> make sure it?s different from previous Message ?
>
> *REGISTER 1*
>
> Via: SIP/2.0/UDP 
> 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
> Via: SIP/2.0/UDP 
> 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
> *REGISTER 2*
>
> Via: SIP/2.0/UDP 
> 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
> Via: SIP/2.0/UDP 
> 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915656050004_MTAuMTIuMy4xMQ
>
> *Thanks*
>
> *Rahul Gupta***
>
> Systems Architect
>
> *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com <mailto:rahul.gupta at ipc.com>
>
> cid:image006.jpg at 01D1940F.3E021840
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Thursday, July 07, 2016 4:43 AM
> *To:* Gupta, Rahul
> *Cc:* Elliott, Ray; users at lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
> Hi Rahul,
>
> The received VIA hdr (with the .44 IP) is properly preserved when 
> forwarding (in the outbound message).
>
> The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and 
> its branch value is completely independent from the branch of other 
> VIA hdrs.
>
> Why do they look the same ? The RFC3261 says that for stateless fwd 
> (when basically there is no transaction, so no branch value), to avoid 
> populating its VIA with ";branch=0" , the proxy may "copy" and use a 
> branch value from an older VIA (a received VIA) - keep in mind it does 
> not say to copy it entirely or part. So, OpenSIPS copies a 
> MAX_BRANCH_PARAM_LEN length string from the previous branch param.
>
> Everything is correct and legal (from RFC perspective).
>
> PS: if you would use t_relay() instead of forward() - doing statefull 
> proxy -, you will see that the branch in the VIA added by OpenSIPS 
> will be completly different from the value in the previous VIA.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 07.07.2016 03:42, Gupta, Rahul wrote:
>
>     Hi Bogdan,
>
>     Here is a Ethernet trace (pcap) file that has a successful and an
>     unsuccessful registration.
>
>     Frames 1-8 illustrate the successful case with Frames 2 and 6 show
>     how opensips adds an extra VIA header that has a full VIA;branch
>
>     Frame:2
>
>     Via: SIP/2.0/UDP
>     10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>     Via: SIP/2.0/UDP
>     10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>     Frames 9-16 illustrate the unsuccessful case where Frames 10 and
>     14 show how opensips adds an extra VIA header that has a truncated
>     branch.
>
>     Frame 10:
>
>     Via: SIP/2.0/UDP
>     10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
>     Via: SIP/2.0/UDP
>     10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
>     *Thanks*
>
>     *Rahul Gupta*
>
>     Systems Architect
>
>     *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com
>     <mailto:rahul.gupta at ipc.com>
>
>     cid:image006.jpg at 01D1940F.3E021840
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Wednesday, July 06, 2016 5:38 AM
>     *To:* OpenSIPS users mailling list
>     *Cc:* Elliott, Ray; Gupta, Rahul
>     *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
>     Hi Rahul,
>
>     That define is used to calculate the the max VIA len when OpenSIPS
>     is generating its own VIA headers. That max len does not impact
>     the VIA headers which were received.
>
>     I do not understand exactly (in your example) what's the flow of
>     that VIA header. If you want, send me off-list the pcap/ngrep
>     showing the SIP package (before and after OpenSIPS) and how it is
>     affected.
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 05.07.2016 20:57, Gupta, Rahul wrote:
>
>         We are using opensips 1.11.5 as a stateless proxy and seeing a
>         truncated Branch in Via for my REGISTER message. After some
>         code digging, the MAX length is calculated using the following
>         formula and seems like its truncating the branch after 55
>         characters. This is causing the REGISTER to fail in our case.
>         Is there a config level solution to this ?
>
>         *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ +
>         1 /*sep*/ + \*
>
>         *MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \*
>
>         *1 /*extra space, needed by t_calc_branch*/)*
>
>         truncated from opensips ?Via: SIP/2.0/UDP
>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
>
>
>         Via from EndPoint ?Via: SIP/2.0/UDP
>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ
>
>
>         *Thanks*
>
>         *Rahul Gupta*
>
>         DISCLAIMER: This e-mail may contain information that is
>         confidential, privileged or otherwise protected from
>         disclosure. If you are not an intended recipient of this
>         e-mail, do not duplicate or redistribute it by any means.
>         Please delete it and any attachments and notify the sender
>         that you have received it in error. Unintended recipients are
>         prohibited from taking action on the basis of information in
>         this e-mail. E-mail messages may contain computer viruses or
>         other defects, may not be accurately replicated on other
>         systems, or may be intercepted, deleted or interfered with
>         without the knowledge of the sender or the intended recipient.
>         If you are not comfortable with the risks associated with
>         e-mail messages, you may decide not to use e-mail to
>         communicate with IPC. IPC reserves the right, to the extent
>         and under circumstances permitted by applicable law, to
>         retain, monitor and intercept e-mail messages to and from its
>         systems.
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From pimenta at inatel.br  Fri Jul  8 21:43:18 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Fri, 8 Jul 2016 19:43:18 +0000
Subject: [OpenSIPS-Users] OpenSIPS and Internet of Things. Where to find
	examples.
Message-ID: <BY2PR02MB165574ADE29E6C0360016646B53C0@BY2PR02MB1655.namprd02.prod.outlook.com>

Dear OpenSIPS users;


This year, in November, I will teach classes about Internet of Things, SIP, SIP Proxy and messaging systems, at Inatel.

I would like to speak about the use of SIP and SIP Proxies in projects related to Internet of Things. But, I am beginning right now looking for the use of SIP in Internet of Things and possibly the use of SIP Proxies in such context.


Does someone here could point me some web site or youtube video that shows any example of using OpenSIPS or any SIP Proxy in an Internet of Things context, please? For example, some article, or site, or document that shows what is the future of SIP in Internet of Things or why a SIP Proxy is useful in IoT will be very appreciated.


Any indication will be very helpful!


Thanks a lot.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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From chandan.pr at webshar.org  Sun Jul 10 10:43:29 2016
From: chandan.pr at webshar.org (Chandan PR)
Date: Sun, 10 Jul 2016 14:13:29 +0530
Subject: [OpenSIPS-Users] OpenSips Reply logged 4 times and not sent back to
	originator
Message-ID: <CADh=FyjF+uZdoPRXJ71BFicDaO1kgk65eUY=-39shX=d4L0d_w@mail.gmail.com>

Hi Guys,

Our Load Balancer configuration is as below:

route{
if (!mf_process_maxfwd_header("3")) {
send_reply("483","looping");
exit;
}


if ( has_totag() ) {
# sequential request -> obey Route indication
loose_route();
                t_relay();
                exit;
        }

        # handle cancel and re-transmissions
if ( is_method("CANCEL") ) {
if ( t_check_trans() )
t_relay();
exit;
}


        # from now on we have only the initial requests
        if (!is_method("INVITE")) {
                send_reply("405","Method Not Allowed");
                exit;
        }

        # initial request
record_route();
# LB function returns negative if no suitable destination (for requested
resources) is found,
        # or if all destinations are full
        if ( !load_balance("1","channel") ) {
             send_reply("500","Service full");
             exit;
        }
t_on_reply("test");
        # arm a failure route for be able to catch a failure event and to
do
        # failover to the next available destination
        t_on_failure("LB_failed");

        # send it out
if (!t_relay()) {
sl_reply_error();
}

}

onreply_route[test] {
xlog("Reply $rs received from $si for the call $avp(iurl)");
}

In the opensips.log file, for some of the calls I see the reply 200 logged
4 times. And whenever that happens, the same is not sent back to the
originator of the call.

At the end of the day some calls are ending up in ambiguous result codes as
they don't receive 200 OK. And all the calls in this category, we see the
reply 200 logged 4 times in the log.

Any clue on what could be the issue?


Regards,
Chandan
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From Ray.Elliott at ipc.com  Sat Jul  9 15:47:01 2016
From: Ray.Elliott at ipc.com (Elliott, Ray)
Date: Sat, 9 Jul 2016 13:47:01 +0000
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <577E1629.6070209@opensips.org>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
Message-ID: <BL2PR15MB1074A011C26D98B286A9396AE83D0@BL2PR15MB1074.namprd15.prod.outlook.com>

Hi Bogdan,

I'm a bit confused.


From bogdan at opensips.org  Mon Jul 11 12:11:44 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jul 2016 13:11:44 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <BL2PR15MB1074A011C26D98B286A9396AE83D0@BL2PR15MB1074.namprd15.prod.outlook.com>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<BL2PR15MB1074A011C26D98B286A9396AE83D0@BL2PR15MB1074.namprd15.prod.outlook.com>
Message-ID: <578370E0.3030301@opensips.org>

Hi Elliott,

1) regarding the uniqueness - it is across time and space ,per server! 
The transaction IDs are per server, so there is not problem if 2 servers 
do use the same ID for a transaction - each server will look into the 
branch ID belonging to its VIA hdr.
So, I do not see any conflict here

2) I never statethat the idea to reuse any previous branch param (to 
build our own VIA branch) comes from RFC - this approach was the result 
of a discussion on the mailing list between developers.

Let me know if I manage to answer to your questions.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 09.07.2016 16:47, Elliott, Ray wrote:
>
> Hi Bogdan,
>
> I?m a bit confused.
>
> From RFC 3261 states in Section 16.11 Stateless Proxy
>
> */The requirement for unique branch IDs across space and time applies 
> to stateless proxies as well. /*
>
> /However, a stateless proxy cannot simply use a random number 
> generator to compute the first component of the branch ID, as 
> described in Section 16.6 bullet 8. This is because retransmissions of 
> a request need to have the same value, and a stateless proxy cannot 
> tell a retransmission from the original request./
>
> */Therefore, the component of the branch parameter that makes it 
> unique MUST be the same each time a retransmitted request is forwarded./*
>
> /Thus for a stateless proxy, the branch parameter MUST be computed as 
> a combinatoric function of message parameters which are invariant on 
> retransmission./
>
> //
>
> We are using the stateless implementation of opensips. As such we MUST 
> not use the statefull  t_replay() function.
>
> Copying the TOP most Via header branch will meet the requirement since 
> that branch value MUST also be ?unique branch IDs across space and time.?
>
> However, just copying apart of that  destroys the ?unique branch IDs 
> across space and time ? requirement.
>
> I?m confused because I can not find the requirement that you specified 
> below in RFC 3261.
>
> Thanks for your response.
>
> Ray
>
> *Ray Elliott*
>
> Principal Scientist
>
> 777, Commerce Dr., Fairfield, CT 06825
>
> *T *+1 203 339 7052  M +1 203 848 4550 *E* ray.elliott at ipc.com 
> <mailto:ray.elliott at ipc.com>
>
> Follow us on twitter: @ipc_Systems_Inc www.ipc.com <http://www.ipc.com/>
>
> cid:image001.jpg at 01D19553.88C2B160
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Thursday, July 07, 2016 4:43 AM
> *To:* Gupta, Rahul
> *Cc:* Elliott, Ray; users at lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
> Hi Rahul,
>
> The received VIA hdr (with the .44 IP) is properly preserved when 
> forwarding (in the outbound message).
>
> The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and 
> its branch value is completely independent from the branch of other 
> VIA hdrs.
>
> Why do they look the same ? The RFC3261 says that for stateless fwd 
> (when basically there is no transaction, so no branch value), to avoid 
> populating its VIA with ";branch=0" , the proxy may "copy" and use a 
> branch value from an older VIA (a received VIA) - keep in mind it does 
> not say to copy it entirely or part. So, OpenSIPS copies a 
> MAX_BRANCH_PARAM_LEN length string from the previous branch param.
>
> Everything is correct and legal (from RFC perspective).
>
> PS: if you would use t_relay() instead of forward() - doing statefull 
> proxy -, you will see that the branch in the VIA added by OpenSIPS 
> will be completly different from the value in the previous VIA.
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 07.07.2016 03:42, Gupta, Rahul wrote:
>
>     Hi Bogdan,
>
>     Here is a Ethernet trace (pcap) file that has a successful and an
>     unsuccessful registration.
>
>     Frames 1-8 illustrate the successful case with Frames 2 and 6 show
>     how opensips adds an extra VIA header that has a full VIA;branch
>
>     Frame:2
>
>     Via: SIP/2.0/UDP
>     10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>     Via: SIP/2.0/UDP
>     10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>     Frames 9-16 illustrate the unsuccessful case where Frames 10 and
>     14 show how opensips adds an extra VIA header that has a truncated
>     branch.
>
>     Frame 10:
>
>     Via: SIP/2.0/UDP
>     10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
>     Via: SIP/2.0/UDP
>     10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
>     *Thanks*
>
>     *Rahul Gupta*
>
>     Systems Architect
>
>     *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com
>     <mailto:rahul.gupta at ipc.com>
>
>     cid:image006.jpg at 01D1940F.3E021840
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Wednesday, July 06, 2016 5:38 AM
>     *To:* OpenSIPS users mailling list
>     *Cc:* Elliott, Ray; Gupta, Rahul
>     *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
>     Hi Rahul,
>
>     That define is used to calculate the the max VIA len when OpenSIPS
>     is generating its own VIA headers. That max len does not impact
>     the VIA headers which were received.
>
>     I do not understand exactly (in your example) what's the flow of
>     that VIA header. If you want, send me off-list the pcap/ngrep
>     showing the SIP package (before and after OpenSIPS) and how it is
>     affected.
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 05.07.2016 20:57, Gupta, Rahul wrote:
>
>         We are using opensips 1.11.5 as a stateless proxy and seeing a
>         truncated Branch in Via for my REGISTER message. After some
>         code digging, the MAX length is calculated using the following
>         formula and seems like its truncating the branch after 55
>         characters. This is causing the REGISTER to fail in our case.
>         Is there a config level solution to this ?
>
>         *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ +
>         1 /*sep*/ + \*
>
>         *MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \*
>
>         *1 /*extra space, needed by t_calc_branch*/)*
>
>         truncated from opensips ?Via: SIP/2.0/UDP
>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
>
>
>         Via from EndPoint ?Via: SIP/2.0/UDP
>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ
>
>
>         *Thanks*
>
>         *Rahul Gupta*
>
>         DISCLAIMER: This e-mail may contain information that is
>         confidential, privileged or otherwise protected from
>         disclosure. If you are not an intended recipient of this
>         e-mail, do not duplicate or redistribute it by any means.
>         Please delete it and any attachments and notify the sender
>         that you have received it in error. Unintended recipients are
>         prohibited from taking action on the basis of information in
>         this e-mail. E-mail messages may contain computer viruses or
>         other defects, may not be accurately replicated on other
>         systems, or may be intercepted, deleted or interfered with
>         without the knowledge of the sender or the intended recipient.
>         If you are not comfortable with the risks associated with
>         e-mail messages, you may decide not to use e-mail to
>         communicate with IPC. IPC reserves the right, to the extent
>         and under circumstances permitted by applicable law, to
>         retain, monitor and intercept e-mail messages to and from its
>         systems.
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From siju.irs at gmail.com  Mon Jul 11 12:14:16 2016
From: siju.irs at gmail.com (Siju Nair)
Date: Mon, 11 Jul 2016 15:44:16 +0530
Subject: [OpenSIPS-Users] Fwd:  Working features in opensips
In-Reply-To: <5A2991EA-5858-4F75-8DF6-66CFCB6A8D2E@gmail.com>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
	<577E24E1.1070406@opensips.org>
	<5A2991EA-5858-4F75-8DF6-66CFCB6A8D2E@gmail.com>
Message-ID: <CAETdsxiQ=YB=j=_k4AB9UoXOmOYAvQoheAWWs4u7o2UTGGm0vA@mail.gmail.com>

Hi Sir

Thank you very much sir for your quick reply.. Can you please sent me a
link of documents to achieve those features, since am unable to find one.

Thanks,
Siju Nair

Sent from my iPhone

> On 07-Jul-2016, at 3:16 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:
>
> Bogdan-Andrei Iancu
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From bogdan at opensips.org  Mon Jul 11 12:24:16 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jul 2016 13:24:16 +0300
Subject: [OpenSIPS-Users] opensip proxy behind firwall
In-Reply-To: <00fd01d1d92c$8f89dba0$ae9d92e0$@bose@rebaca.com>
References: <00fd01d1d92c$8f89dba0$ae9d92e0$@bose@rebaca.com>
Message-ID: <578373D0.6020303@opensips.org>

Hi,

Could you be more specific in what the "fail to make calls" means ?

Thanks,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 08.07.2016 18:22, Priyanko Bose wrote:
>
> Hi,
>
> I am trying to setup a OpenSIP server which should act as a SIP 
> register as well as a SIP proxy server.
>
> If the server is placed in the same LAN where multiple SIP enabled 
> devices (e.g Cisco SX20, Polycom RPG500) reside, then the devices  get 
> registered  easily and can make SIP calls among themselves.
>
> But everything fails when the devices are on the other side of the 
> internet and the server is behind firewall like following diagram.
>
> ----------     ------------ 
>                                  -------------             --------------
>
> - Device -    <--->       -    F/W -              <--->  
>                      -  F/W -                     <--->  
>                   - OpenSIPS  -
>
> ----------       ------------ 
>                                      -------------            
>  --------------
>
> 192.168.v.w(LAN)    192.168.v.1 : 63.45.x.y     (WAN)   220.54.w.y: 
> 192.168.x.1       (LAN) 192.168.x.y
>
> The server?s private IP (192.168.x.y ) is mapped to a fixed public IP 
> (220.54.w.y) and ?opensips.cfg? file has been prepared same as 
> described in the link 
> http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Server-behind-a-NAT-firewall-td5853352.html. 
> This time I am able to register the device but fails to make SIP call.
>
> Regards,
>
> Priyanko
>

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From bogdan at opensips.org  Mon Jul 11 12:27:21 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jul 2016 13:27:21 +0300
Subject: [OpenSIPS-Users] OpenSips Reply logged 4 times and not sent
 back to originator
In-Reply-To: <CADh=FyjF+uZdoPRXJ71BFicDaO1kgk65eUY=-39shX=d4L0d_w@mail.gmail.com>
References: <CADh=FyjF+uZdoPRXJ71BFicDaO1kgk65eUY=-39shX=d4L0d_w@mail.gmail.com>
Message-ID: <57837489.4050504@opensips.org>

Hi,

It is hard to say without a pcap - try to make a call capturing on the 
LB server, showing both in and out SIP packets.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.07.2016 11:43, Chandan PR wrote:
> Hi Guys,
>
> Our Load Balancer configuration is as below:
>
> route{
> if (!mf_process_maxfwd_header("3")) {
> send_reply("483","looping");
> exit;
> }
>
>
> if ( has_totag() ) {
> # sequential request -> obey Route indication
> loose_route();
>                 t_relay();
>                 exit;
>         }
>
>         # handle cancel and re-transmissions
> if ( is_method("CANCEL") ) {
> if ( t_check_trans() )
> t_relay();
> exit;
> }
>
>
>         # from now on we have only the initial requests
>         if (!is_method("INVITE")) {
>                 send_reply("405","Method Not Allowed");
>                 exit;
>         }
>
>         # initial request
> record_route();
> # LB function returns negative if no suitable destination (for 
> requested resources) is found,
>         # or if all destinations are full
>         if ( !load_balance("1","channel") ) {
>              send_reply("500","Service full");
>              exit;
>         }
> t_on_reply("test");
>         # arm a failure route for be able to catch a failure event and 
> to do
>         # failover to the next available destination
>         t_on_failure("LB_failed");
>
>         # send it out
> if (!t_relay()) {
> sl_reply_error();
> }
>
> }
>
> onreply_route[test] {
> xlog("Reply $rs received from $si for the call $avp(iurl)");
> }
>
> In the opensips.log file, for some of the calls I see the reply 200 
> logged 4 times. And whenever that happens, the same is not sent back 
> to the originator of the call.
>
> At the end of the day some calls are ending up in ambiguous result 
> codes as they don't receive 200 OK. And all the calls in this 
> category, we see the reply 200 logged 4 times in the log.
>
> Any clue on what could be the issue?
>
>
> Regards,
> Chandan
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Mon Jul 11 12:28:46 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jul 2016 13:28:46 +0300
Subject: [OpenSIPS-Users] Fwd:  Working features in opensips
In-Reply-To: <CAETdsxiQ=YB=j=_k4AB9UoXOmOYAvQoheAWWs4u7o2UTGGm0vA@mail.gmail.com>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
	<577E24E1.1070406@opensips.org>
	<5A2991EA-5858-4F75-8DF6-66CFCB6A8D2E@gmail.com>
	<CAETdsxiQ=YB=j=_k4AB9UoXOmOYAvQoheAWWs4u7o2UTGGm0vA@mail.gmail.com>
Message-ID: <578374DE.8070804@opensips.org>

Hi,

The features are implement by doing OpenSIPS scripting (based on how 
such services should work in SIP). There is no explicit documentation on 
how to implement something like this.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.07.2016 13:14, Siju Nair wrote:
> Hi Sir
>
> Thank you very much sir for your quick reply.. Can you please sent me 
> a link of documents to achieve those features, since am unable to find 
> one.
>
> Thanks,
> Siju Nair
>
> Sent from my iPhone
>
> > On 07-Jul-2016, at 3:16 PM, Bogdan-Andrei Iancu <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
> >
> > Bogdan-Andrei Iancu
>

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From siju.irs at gmail.com  Mon Jul 11 14:18:01 2016
From: siju.irs at gmail.com (Siju Nair)
Date: Mon, 11 Jul 2016 17:48:01 +0530
Subject: [OpenSIPS-Users] Fwd:  Working features in opensips
In-Reply-To: <578374DE.8070804@opensips.org>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
	<577E24E1.1070406@opensips.org>
	<5A2991EA-5858-4F75-8DF6-66CFCB6A8D2E@gmail.com>
	<CAETdsxiQ=YB=j=_k4AB9UoXOmOYAvQoheAWWs4u7o2UTGGm0vA@mail.gmail.com>
	<578374DE.8070804@opensips.org>
Message-ID: <CAETdsxihcUagOQjQsFOsG7hKm7qAK_xb5en-7zTUhxsq=svfqw@mail.gmail.com>

Hi Bogdan,
Thank you very much for your reply, so can you help me with below scenario,

-> i will be doing all sip registrations on Opensips,which is fine, but i
wanted all type of calls (like local, trunk pstn) should be handled through
media servers like Asterisk/freeswitch, is that possible.?

On Mon, Jul 11, 2016 at 3:58 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi,
>
> The features are implement by doing OpenSIPS scripting (based on how such
> services should work in SIP). There is no explicit documentation on how to
> implement something like this.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 11.07.2016 13:14, Siju Nair wrote:
>
> Hi Sir
>
> Thank you very much sir for your quick reply.. Can you please sent me a
> link of documents to achieve those features, since am unable to find one.
>
> Thanks,
> Siju Nair
>
> Sent from my iPhone
>
> > On 07-Jul-2016, at 3:16 PM, Bogdan-Andrei Iancu < <bogdan at opensips.org>
> bogdan at opensips.org> wrote:
> >
> > Bogdan-Andrei Iancu
>
>
>
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From annusfictus at gmail.com  Mon Jul 11 15:45:23 2016
From: annusfictus at gmail.com (Annus Fictus)
Date: Mon, 11 Jul 2016 15:45:23 +0200
Subject: [OpenSIPS-Users] Fwd: Working features in opensips
In-Reply-To: <CAETdsxihcUagOQjQsFOsG7hKm7qAK_xb5en-7zTUhxsq=svfqw@mail.gmail.com>
References: <CAETdsxgoMmN2A1Xy5VZM-55UohvutJZa_OVP9yTrN0Lk0doH+w@mail.gmail.com>
	<577E24E1.1070406@opensips.org>
	<5A2991EA-5858-4F75-8DF6-66CFCB6A8D2E@gmail.com>
	<CAETdsxiQ=YB=j=_k4AB9UoXOmOYAvQoheAWWs4u7o2UTGGm0vA@mail.gmail.com>
	<578374DE.8070804@opensips.org>
	<CAETdsxihcUagOQjQsFOsG7hKm7qAK_xb5en-7zTUhxsq=svfqw@mail.gmail.com>
Message-ID: <7dd969bf-25e7-b3da-3779-80f0ec7b5fcf@gmail.com>

Hello,

take a look to LOAD_BALANCER Module.

http://www.opensips.org/html/docs/modules/1.11.x/load_balancer.html

Regards

El 11/07/2016 a las 14:18, Siju Nair escribi?:
> Hi Bogdan,
> Thank you very much for your reply, so can you help me with below 
> scenario,
>
> -> i will be doing all sip registrations on Opensips,which is fine, 
> but i wanted all type of calls (like local, trunk pstn) should be 
> handled through media servers like Asterisk/freeswitch, is that 
> possible.?
>
> On Mon, Jul 11, 2016 at 3:58 PM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi,
>
>     The features are implement by doing OpenSIPS scripting (based on
>     how such services should work in SIP). There is no explicit
>     documentation on how to implement something like this.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 11.07.2016 13:14, Siju Nair wrote:
>>     Hi Sir
>>
>>     Thank you very much sir for your quick reply.. Can you please
>>     sent me a link of documents to achieve those features, since am
>>     unable to find one.
>>
>>     Thanks,
>>     Siju Nair
>>
>>     Sent from my iPhone
>>
>>     > On 07-Jul-2016, at 3:16 PM, Bogdan-Andrei Iancu
>>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>     >
>>     > Bogdan-Andrei Iancu
>>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Mon Jul 11 18:42:19 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jul 2016 19:42:19 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <577FB917.6020400@opensips.org>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577FB917.6020400@opensips.org>
Message-ID: <5783CC6B.6040608@opensips.org>

Hi Rahul,

There is a fix available for extra testing - see :
https://github.com/OpenSIPS/opensips/commit/482e643469b351d12418ff54c96beee7b27dca94

please give it a try and let me know if it is solving the problem for 
you too.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 08.07.2016 17:30, Bogdan-Andrei Iancu wrote:
> Hi Rahul,
>
> Indeed, that is a good pointhere, thanks for bring it up. Let me 
> investigate the code a bit and I will update you.
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 07.07.2016 22:01, Gupta, Rahul wrote:
>>
>> Hi Bogdan, so the problem we are facing is the endpoint has a new 
>> branch however the difference in the branch value is *AFTER* the 
>> MAX_BRANCH_PARAM_LEN, the highlighted text.Since opensips is copying 
>> the partial branch which is same as the previous Via and thus our SIP 
>> stack (OCCAS) is sending back 401 Unauthorized for both thinking it?s 
>> a retransmit.
>>
>> This is also in line with RFC3261. Shouldn?t it copy the entire Via 
>> to make sure it?s different from previous Message ?
>>
>> *REGISTER 1*
>>
>> Via: SIP/2.0/UDP 
>> 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>>
>> Via: SIP/2.0/UDP 
>> 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>>
>> *REGISTER 2*
>>
>> Via: SIP/2.0/UDP 
>> 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>>
>> Via: SIP/2.0/UDP 
>> 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915656050004_MTAuMTIuMy4xMQ
>>
>> *Thanks*
>>
>> *Rahul Gupta***
>>
>> Systems Architect
>>
>> *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com <mailto:rahul.gupta at ipc.com>
>>
>> cid:image006.jpg at 01D1940F.3E021840
>>
>> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>> *Sent:* Thursday, July 07, 2016 4:43 AM
>> *To:* Gupta, Rahul
>> *Cc:* Elliott, Ray; users at lists.opensips.org
>> *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>>
>> Hi Rahul,
>>
>> The received VIA hdr (with the .44 IP) is properly preserved when 
>> forwarding (in the outbound message).
>>
>> The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and 
>> its branch value is completely independent from the branch of other 
>> VIA hdrs.
>>
>> Why do they look the same ? The RFC3261 says that for stateless fwd 
>> (when basically there is no transaction, so no branch value), to 
>> avoid populating its VIA with ";branch=0" , the proxy may "copy" and 
>> use a branch value from an older VIA (a received VIA) - keep in mind 
>> it does not say to copy it entirely or part. So, OpenSIPS copies a 
>> MAX_BRANCH_PARAM_LEN length string from the previous branch param.
>>
>> Everything is correct and legal (from RFC perspective).
>>
>> PS: if you would use t_relay() instead of forward() - doing statefull 
>> proxy -, you will see that the branch in the VIA added by OpenSIPS 
>> will be completly different from the value in the previous VIA.
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>> On 07.07.2016 03:42, Gupta, Rahul wrote:
>>
>>     Hi Bogdan,
>>
>>     Here is a Ethernet trace (pcap) file that has a successful and an
>>     unsuccessful registration.
>>
>>     Frames 1-8 illustrate the successful case with Frames 2 and 6
>>     show how opensips adds an extra VIA header that has a full VIA;branch
>>
>>     Frame:2
>>
>>     Via: SIP/2.0/UDP
>>     10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>>
>>     Via: SIP/2.0/UDP
>>     10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>>
>>     Frames 9-16 illustrate the unsuccessful case where Frames 10 and
>>     14 show how opensips adds an extra VIA header that has a
>>     truncated branch.
>>
>>     Frame 10:
>>
>>     Via: SIP/2.0/UDP
>>     10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>>
>>     Via: SIP/2.0/UDP
>>     10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>>
>>     *Thanks*
>>
>>     *Rahul Gupta*
>>
>>     Systems Architect
>>
>>     *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com
>>     <mailto:rahul.gupta at ipc.com>
>>
>>     cid:image006.jpg at 01D1940F.3E021840
>>
>>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>>     *Sent:* Wednesday, July 06, 2016 5:38 AM
>>     *To:* OpenSIPS users mailling list
>>     *Cc:* Elliott, Ray; Gupta, Rahul
>>     *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>>
>>     Hi Rahul,
>>
>>     That define is used to calculate the the max VIA len when
>>     OpenSIPS is generating its own VIA headers. That max len does not
>>     impact the VIA headers which were received.
>>
>>     I do not understand exactly (in your example) what's the flow of
>>     that VIA header. If you want, send me off-list the pcap/ngrep
>>     showing the SIP package (before and after OpenSIPS) and how it is
>>     affected.
>>
>>     Regards,
>>
>>
>>     Bogdan-Andrei Iancu
>>
>>     OpenSIPS Founder and Developer
>>
>>     http://www.opensips-solutions.com
>>
>>     On 05.07.2016 20:57, Gupta, Rahul wrote:
>>
>>         We are using opensips 1.11.5 as a stateless proxy and seeing
>>         a truncated Branch in Via for my REGISTER message. After some
>>         code digging, the MAX length is calculated using the
>>         following formula and seems like its truncating the branch
>>         after 55 characters. This is causing the REGISTER to fail in
>>         our case. Is there a config level solution to this ?
>>
>>         *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ +
>>         1 /*sep*/ + \*
>>
>>         *MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \*
>>
>>         *1 /*extra space, needed by t_calc_branch*/)*
>>
>>         truncated from opensips ?Via: SIP/2.0/UDP
>>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
>>
>>
>>         Via from EndPoint ?Via: SIP/2.0/UDP
>>         XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ
>>
>>
>>         *Thanks*
>>
>>         *Rahul Gupta*
>>
>>         DISCLAIMER: This e-mail may contain information that is
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>>
>>
>>
>>
>>
>>         _______________________________________________
>>
>>         Users mailing list
>>
>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From johan at democon.be  Tue Jul 12 08:27:08 2016
From: johan at democon.be (Johan De Clercq)
Date: Tue, 12 Jul 2016 08:27:08 +0200
Subject: [OpenSIPS-Users] rtpengine trouble.
Message-ID: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>

Please find attached

- opensips.cfg,

- opensips.log (rtpengine dbg output only)

- rtpengine (init.d)


pastebin:(http://pastebin.com/ct8rCEwE)

- syslog (contains opensips dbg output )



Problem : opensips can?t communicate with rtpengine. (used to work with
1.11 and 2.1 )

Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly parse UDP
command line '2470_0 d7:command4:pinge' from 82.199.64.147:60185, using
fallback RE



Opensips is 2.2 from git; rtpengine is latest from git (4.0).



Can you please indicate what my problem is or please indicate what logs you
need ?


Br, Johan.
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From linuxvenkey at gmail.com  Wed Jul 13 11:52:49 2016
From: linuxvenkey at gmail.com (Venkatesh Macha)
Date: Wed, 13 Jul 2016 02:52:49 -0700 (MST)
Subject: [OpenSIPS-Users] how to Call center module Agent location update
Message-ID: <1468403569925-7603772.post@n2.nabble.com>

Hi all,

   I am using OpenSIPS Call Center module, I am able to configure it and It
is working fine. But I have one question.
Is there any way to automatically update the* cc_agents* table *location*
filed ?

Now I am manually adding the agent to the *cc_agents *table. It is working
as expected. But if the agent may log in from different devices, So it is
good idea to update this on the Registration message. 

I also tried using *cc_agent_login("agentX","1")*, But it is not updating
the location field. Yes agent is login is successful.

Is there any way to update the* cc_agents* table *location* field on the
registration ?

I am using OpenSIPS devel branch.

Thanks & Regards,
Venkatesh Macha.
 



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-Call-center-module-Agent-location-update-tp7603772.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


From feroze.chaudhry at gmail.com  Wed Jul 13 15:44:44 2016
From: feroze.chaudhry at gmail.com (feroze waris)
Date: Wed, 13 Jul 2016 18:44:44 +0500
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
Message-ID: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>

Hi,

Can anyone tell me how to move opensips DB from SQL to NoSQL. I have seen a
module name db_cachedb which has a support for mongodb but how to move
mysql schema to mongodb schema.

Regards
Feroze
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From daniel.zanutti at gmail.com  Wed Jul 13 15:52:07 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Wed, 13 Jul 2016 10:52:07 -0300
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
Message-ID: <CAN0jgiNGbs-rBwvmaQFy+J_qT5fGyGWaHxLcnk98uGhjSPk7=g@mail.gmail.com>

Hi Feroze

I think you should first understand exactly what is written to MySQL and
answer yourself why do you want to move to NoSQL.

Opensips has many modules that access DB structures like ACC, Subscribers,
Address, Location, Loadbalancer, cachedb, etc. Which do you want to move to
NoSQL? And why? Is this a performance issue?

Cachedb is for caching results and you can use NoSQL to have a persistent
cache.

Regards


On Wed, Jul 13, 2016 at 10:44 AM, feroze waris <feroze.chaudhry at gmail.com>
wrote:

> Hi,
>
> Can anyone tell me how to move opensips DB from SQL to NoSQL. I have seen
> a module name db_cachedb which has a support for mongodb but how to move
> mysql schema to mongodb schema.
>
> Regards
> Feroze
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From benjamin.cropley at gmail.com  Wed Jul 13 15:52:36 2016
From: benjamin.cropley at gmail.com (Benjamin Cropley)
Date: Wed, 13 Jul 2016 14:52:36 +0100
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
Message-ID: <CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>

You don't have to explicitly define a schema when you insert a document
into MongoDB.

For example, you could create a MongoDB database, and then immediately do
an insert into an insert into a collection without even creating it or a
'schema'.

Maybe you should do some reading about MongoDB and NoSQL in general, as a
basic understanding should have answered that question :)

https://docs.mongodb.com/manual/faq/fundamentals/

On Wed, Jul 13, 2016 at 2:44 PM, feroze waris <feroze.chaudhry at gmail.com>
wrote:

> Hi,
>
> Can anyone tell me how to move opensips DB from SQL to NoSQL. I have seen
> a module name db_cachedb which has a support for mongodb but how to move
> mysql schema to mongodb schema.
>
> Regards
> Feroze
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
All the best,
Ben Cropley
07539 366 905
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From millennium.bug at gmail.com  Wed Jul 13 19:51:45 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Wed, 13 Jul 2016 22:51:45 +0500
Subject: [OpenSIPS-Users] horizontal scaling / dimensioning
Message-ID: <CAArAGKkRngJd4heuZB46-Uny-19aPaiVTNnn4t58ABLApq8cxw@mail.gmail.com>

Hi all,

Is there a way I can theoretically determine the max number of dialogs and
cps a system of known specifications can handle?

Also share useful tools for dimensioning, and your findings of max
concurrent calls and cps on single opensips instances with specs.

Thanks,
Owais
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From eric at uphreak.com  Wed Jul 13 20:09:07 2016
From: eric at uphreak.com (Eric Tamme)
Date: Wed, 13 Jul 2016 12:09:07 -0600
Subject: [OpenSIPS-Users] horizontal scaling / dimensioning
In-Reply-To: <CAArAGKkRngJd4heuZB46-Uny-19aPaiVTNnn4t58ABLApq8cxw@mail.gmail.com>
References: <CAArAGKkRngJd4heuZB46-Uny-19aPaiVTNnn4t58ABLApq8cxw@mail.gmail.com>
Message-ID: <578683C3.1030207@uphreak.com>

Short answer,  no.  There are many variables that will change what a 
system is capable of and without understanding all the pieces involved, 
you can not make any estimate at sizing.

On 07/13/2016 11:51 AM, Owais Ahmad wrote:
> Hi all,
>
> Is there a way I can theoretically determine the max number of dialogs 
> and cps a system of known specifications can handle?
>
> Also share useful tools for dimensioning, and your findings of max 
> concurrent calls and cps on single opensips instances with specs.
>
> Thanks,
> Owais
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From qasimakhan at gmail.com  Wed Jul 13 20:23:18 2016
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Wed, 13 Jul 2016 23:23:18 +0500
Subject: [OpenSIPS-Users] horizontal scaling / dimensioning
In-Reply-To: <578683C3.1030207@uphreak.com>
References: <CAArAGKkRngJd4heuZB46-Uny-19aPaiVTNnn4t58ABLApq8cxw@mail.gmail.com>
	<578683C3.1030207@uphreak.com>
Message-ID: <CAPBKugvmJUDBfN3+C4iYp5j5RrrMxH1ZxYYHY8_0qvNAbdxkpw@mail.gmail.com>

I guess you can reffer to this -> (
http://www.opensips.org/About/PerformanceTests-StressTests)  document for
some estimates and +1 Eric.

Regards,
Qasim

On Wed, Jul 13, 2016 at 11:09 PM, Eric Tamme <eric at uphreak.com> wrote:

> Short answer,  no.  There are many variables that will change what a
> system is capable of and without understanding all the pieces involved, you
> can not make any estimate at sizing.
>
>
> On 07/13/2016 11:51 AM, Owais Ahmad wrote:
>
> Hi all,
>
> Is there a way I can theoretically determine the max number of dialogs and
> cps a system of known specifications can handle?
>
> Also share useful tools for dimensioning, and your findings of max
> concurrent calls and cps on single opensips instances with specs.
>
> Thanks,
> Owais
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From adnan.k at spactron.com  Thu Jul 14 07:44:15 2016
From: adnan.k at spactron.com (Adnan Kurshid)
Date: Thu, 14 Jul 2016 05:44:15 +0000
Subject: [OpenSIPS-Users] Supporting IPv6-only Networks
Message-ID: <AM4PR0701MB22446868B0CF6C97D69E0A5192320@AM4PR0701MB2244.eurprd07.prod.outlook.com>

We are using Opensips version 1.9 and when we tested our iPhone app on IPv6-only network then we are not able to make calls. On IPv4 network everything is working fine. Now apple app review team rejects the app if it is not supporting IPv6-only network. For more details please visit https://developer.apple.com/news/?id=05042016a


Is IPv6-only network supported in any current version of opensips server?


Regards,


Adnan Khurshid

Supporting IPv6-only Networks - News and Updates - Apple ...<https://developer.apple.com/news/?id=05042016a>
developer.apple.com
Get updates, tips, and how-to information on a range of development, App Store, and Mac App Store topics.


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From bogdan at opensips.org  Thu Jul 14 11:16:51 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 12:16:51 +0300
Subject: [OpenSIPS-Users] Supporting IPv6-only Networks
In-Reply-To: <AM4PR0701MB22446868B0CF6C97D69E0A5192320@AM4PR0701MB2244.eurprd07.prod.outlook.com>
References: <AM4PR0701MB22446868B0CF6C97D69E0A5192320@AM4PR0701MB2244.eurprd07.prod.outlook.com>
Message-ID: <57875883.2030409@opensips.org>

Hi Adnan,

OpenSIPS does support IPv6 (including version 1.9, which anyhow is 
really, really old) - I'm quite sure it is a configuration issue on your 
side.

Could you details what exactly is failing in your test ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 08:44, Adnan Kurshid wrote:
>
> We are using Opensips version 1.9 and when we tested our iPhone app on 
> IPv6-only network then we are not able to make calls. On IPv4 network 
> everything is working fine. Now apple app review team rejects the app 
> if it is not supporting IPv6-only network. For more details please 
> visit https://developer.apple.com/news/?id=05042016a 
> <https://developer.apple.com/news/?id=05042016a>
>
>
> Is IPv6-only network supported in any current version of opensips server?
>
>
> Regards,
>
>
> Adnan Khurshid
>
> Supporting IPv6-only Networks - News and Updates - Apple ... 
> <https://developer.apple.com/news/?id=05042016a>
> developer.apple.com
> Get updates, tips, and how-to information on a range of development, 
> App Store, and Mac App Store topics.
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Thu Jul 14 11:25:05 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 12:25:05 +0300
Subject: [OpenSIPS-Users] how to Call center module Agent location update
In-Reply-To: <1468403569925-7603772.post@n2.nabble.com>
References: <1468403569925-7603772.post@n2.nabble.com>
Message-ID: <57875A71.1060004@opensips.org>

Hi Venkatesh,

Note that the cc_agents table is a configuration table (defines the 
agents), while the location table keep runtime, volatile data (the 
subscriber registrations).
If you want to do analogies, the cc_agents may correspond to subscriber 
table (where the SIP subscribers are defined).

Anyhow, what you should do is to statically define the agents (via 
cc_agents table) and you can dynamically link the registration event to 
the CC login event. Use the aor-related routed in usrloc module to 
capture in script the event of a user registering / de-registering and 
use from there the cc_agent_login() function

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 13.07.2016 12:52, Venkatesh Macha wrote:
> Hi all,
>
>     I am using OpenSIPS Call Center module, I am able to configure it and It
> is working fine. But I have one question.
> Is there any way to automatically update the* cc_agents* table *location*
> filed ?
>
> Now I am manually adding the agent to the *cc_agents *table. It is working
> as expected. But if the agent may log in from different devices, So it is
> good idea to update this on the Registration message.
>
> I also tried using *cc_agent_login("agentX","1")*, But it is not updating
> the location field. Yes agent is login is successful.
>
> Is there any way to update the* cc_agents* table *location* field on the
> registration ?
>
> I am using OpenSIPS devel branch.
>
> Thanks & Regards,
> Venkatesh Macha.
>   
>
>
>
> --
> View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-Call-center-module-Agent-location-update-tp7603772.html
> Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From adnan.k at spactron.com  Thu Jul 14 11:27:21 2016
From: adnan.k at spactron.com (Adnan Kurshid)
Date: Thu, 14 Jul 2016 09:27:21 +0000
Subject: [OpenSIPS-Users] Supporting IPv6-only Networks
In-Reply-To: <57875883.2030409@opensips.org>
References: <AM4PR0701MB22446868B0CF6C97D69E0A5192320@AM4PR0701MB2244.eurprd07.prod.outlook.com>,
	<57875883.2030409@opensips.org>
Message-ID: <AM4PR0701MB224418EEFA0BE2B9922BA52F92320@AM4PR0701MB2244.eurprd07.prod.outlook.com>

Dear Bogdan-Andrei,


Thanks for your quick reply. I am behind NAT64 network and I am able to connect to SIP Server but when I make a SIP call then I don't get any response from server (time out). But when I am on IPv4 network then everything works fine. I have enabled

dns_try_ipv6=yes


in the config file.


Following is the details of demo accounts that we are using.
1) 500 at 54.235.173.114 password: 500

2) 600 password 600


Regards,


Adnan Khurshid

________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Thursday, July 14, 2016 2:16:51 PM
To: OpenSIPS users mailling list; Adnan Kurshid
Subject: Re: [OpenSIPS-Users] Supporting IPv6-only Networks

Hi Adnan,

OpenSIPS does support IPv6 (including version 1.9, which anyhow is really, really old) - I'm quite sure it is a configuration issue on your side.

Could you details what exactly is failing in your test ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 08:44, Adnan Kurshid wrote:

We are using Opensips version 1.9 and when we tested our iPhone app on IPv6-only network then we are not able to make calls. On IPv4 network everything is working fine. Now apple app review team rejects the app if it is not supporting IPv6-only network. For more details please visit <https://developer.apple.com/news/?id=05042016a> https://developer.apple.com/news/?id=05042016a


Is IPv6-only network supported in any current version of opensips server?


Regards,


Adnan Khurshid

Supporting IPv6-only Networks - News and Updates - Apple ...<https://developer.apple.com/news/?id=05042016a>
developer.apple.com
Get updates, tips, and how-to information on a range of development, App Store, and Mac App Store topics.





_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From bogdan at opensips.org  Thu Jul 14 11:27:32 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 12:27:32 +0300
Subject: [OpenSIPS-Users] rtpengine trouble.
In-Reply-To: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
References: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
Message-ID: <57875B04.6070202@opensips.org>

Hi Johan,

So, what you say is that the rtpengine 4.0 is compatible with OpenSIPS 
1.11 and 2.1 (using the rtpengine module ??) but not with OpenSIPS 2.2 ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 12.07.2016 09:27, Johan De Clercq wrote:
>
> Please find attached
>
> - opensips.cfg,
>
> - opensips.log (rtpengine dbg output only)
>
> - rtpengine (init.d)
>
>
> pastebin:(http://pastebin.com/ct8rCEwE)
>
> - syslog (contains opensips dbg output )
>
> Problem : opensips can?t communicate with rtpengine. (used to work 
> with 1.11 and 2.1 )
>
> Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly parse UDP 
> command line '2470_0 d7:command4:pinge' from 82.199.64.147:60185 
> <http://82.199.64.147:60185>, using fallback RE
>
> Opensips is 2.2 from git; rtpengine is latest from git (4.0).
>
> Can you please indicate what my problem is or please indicate what 
> logs you need ?
>
> Br, Johan.
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From linuxvenkey at gmail.com  Thu Jul 14 11:29:21 2016
From: linuxvenkey at gmail.com (Venkatesh Macha)
Date: Thu, 14 Jul 2016 02:29:21 -0700 (MST)
Subject: [OpenSIPS-Users] how to Call center module Agent location update
In-Reply-To: <57875A71.1060004@opensips.org>
References: <1468403569925-7603772.post@n2.nabble.com>
	<57875A71.1060004@opensips.org>
Message-ID: <1468488561067-7603785.post@n2.nabble.com>

Thanks, Bogdan, I will try to do it.

Venkatesh Macha.



--
View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/how-to-automatically-update-Call-center-module-Agent-location-tp7603772p7603785.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.


From bogdan at opensips.org  Thu Jul 14 11:29:49 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 12:29:49 +0300
Subject: [OpenSIPS-Users] Supporting IPv6-only Networks
In-Reply-To: <AM4PR0701MB224418EEFA0BE2B9922BA52F92320@AM4PR0701MB2244.eurprd07.prod.outlook.com>
References: <AM4PR0701MB22446868B0CF6C97D69E0A5192320@AM4PR0701MB2244.eurprd07.prod.outlook.com>
	<57875883.2030409@opensips.org>
	<AM4PR0701MB224418EEFA0BE2B9922BA52F92320@AM4PR0701MB2244.eurprd07.prod.outlook.com>
Message-ID: <57875B8D.2000401@opensips.org>

So, OpenSIPS receives the INVITE - do you see any OpenSIPS sending out 
any reply (see logs or local traces on the OpenSIPS machine).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 12:27, Adnan Kurshid wrote:
>
> Dear Bogdan-Andrei,
>
>
> Thanks for your quick reply. I am behind NAT64 network and I am able 
> to connect to SIP Server but when I make a SIP call then I don't get 
> any response from server (time out). But when I am on IPv4 network 
> then everything works fine. I have enabled
>
> dns_try_ipv6=yes
>
>
> in the config file.
>
>
> Following is the details of demo accounts that we are using.
> 1) 500 at 54.235.173.114 password: 500
> 2) 600 password 600
>
>
> Regards,
>
>
> Adnan Khurshid
>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Sent:* Thursday, July 14, 2016 2:16:51 PM
> *To:* OpenSIPS users mailling list; Adnan Kurshid
> *Subject:* Re: [OpenSIPS-Users] Supporting IPv6-only Networks
> Hi Adnan,
>
> OpenSIPS does support IPv6 (including version 1.9, which anyhow is 
> really, really old) - I'm quite sure it is a configuration issue on 
> your side.
>
> Could you details what exactly is failing in your test ?
>
> Regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 14.07.2016 08:44, Adnan Kurshid wrote:
>>
>> We are using Opensips version 1.9 and when we tested our iPhone app 
>> on IPv6-only network then we are not able to make calls. On IPv4 
>> network everything is working fine. Now apple app review team rejects 
>> the app if it is not supporting IPv6-only network. For more details 
>> please visit 
>> <https://developer.apple.com/news/?id=05042016a>https://developer.apple.com/news/?id=05042016a
>>
>>
>> Is IPv6-only network supported in any current version of opensips server?
>>
>>
>> Regards,
>>
>>
>> Adnan Khurshid
>>
>> Supporting IPv6-only Networks - News and Updates - Apple ... 
>> <https://developer.apple.com/news/?id=05042016a>
>> developer.apple.com
>> Get updates, tips, and how-to information on a range of development, 
>> App Store, and Mac App Store topics.
>>
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From dujinfang at gmail.com  Thu Jul 14 11:31:04 2016
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 14 Jul 2016 17:31:04 +0800
Subject: [OpenSIPS-Users] rtpengine trouble.
In-Reply-To: <57875B04.6070202@opensips.org>
References: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
	<57875B04.6070202@opensips.org>
Message-ID: <40ABEC79E4864D9A85146F88A4B2E1BE@gmail.com>

FYI, I installed opensips 2.2 from debian 8 repo and rtpengine from git master a few days ago seems worked.  

--  
Seven Du
Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Thursday, July 14, 2016 at 5:27 PM, Bogdan-Andrei Iancu wrote:

> Hi Johan,
>  
> So, what you say is that the rtpengine 4.0 is compatible with OpenSIPS 1.11 and 2.1 (using the rtpengine module ??) but not with OpenSIPS 2.2 ?
>  
> Regards,
> Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com  
> On 12.07.2016 09:27, Johan De Clercq wrote:
> >    
> > Please find attached
> > - opensips.cfg,  
> > - opensips.log (rtpengine dbg output only)
> > - rtpengine (init.d)  
> >  
> > pastebin:(http://pastebin.com/ct8rCEwE)
> > - syslog (contains opensips dbg output )  
> >   
> > Problem : opensips can?t communicate with rtpengine. (used to work with 1.11 and 2.1 )
> > Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly parse UDP command line '2470_0 d7:command4:pinge' from 82.199.64.147:60185 (http://82.199.64.147:60185), using fallback RE
> >   
> > Opensips is 2.2 from git; rtpengine is latest from git (4.0).  
> >   
> > Can you please indicate what my problem is or please indicate what logs you need ?
> >   
> > Br, Johan.  
> >  
> >  
> > _______________________________________________ Users mailing list Users at lists.opensips.org (mailto:Users at lists.opensips.org) http://lists.opensips.org/cgi-bin/mailman/listinfo/users  
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org (mailto:Users at lists.opensips.org)
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  
>  


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From bogdan at opensips.org  Thu Jul 14 12:09:45 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 13:09:45 +0300
Subject: [OpenSIPS-Users] Question regarding b2b_bridge function
In-Reply-To: <CAOooq9BceX-x4qkQKXGJRyamrgqVkRKCXfQ1kNwO=LP4ZNiMPg@mail.gmail.com>
References: <CAOooq9CTERn3oax+TKWsAium7MbLeGvT2dNssWHkrJfgvo6bEg@mail.gmail.com>
	<CAOooq9BY9=osAS8OAMB4GP1z50LUR37WJrCMU3Hn9_Fnk=eTXw@mail.gmail.com>
	<574D44F7.9090204@opensips.org>
	<CAOooq9DHkdbs=XGvwzGMXzZ2hO5_K1TfJeDRkkY6ufa+s0OJQg@mail.gmail.com>
	<574E8B1A.7070508@opensips.org>
	<CAOooq9AzjYHA3GuhiHbfJPj2z30QU8Vhsq1NdUwH+DHtbQFXvA@mail.gmail.com>
	<574FE1C0.9000904@opensips.org>
	<CAOooq9Du1PoEHLdvGEdMZWE1k5rUWqif_ZwKB8VH07Pv+ixEgg@mail.gmail.com>
	<575145C9.9080406@opensips.org>
	<CAOooq9DRtvPTQstPEe56HBq2KOBYVOiych3QoUWRBCw44m6UbQ@mail.gmail.com>
	<575583A1.7050501@opensips.org>
	<CAOooq9DZCUWWupDKkZxp5b1sQz=gBzBCmNnytyyw_FKVMb_67A@mail.gmail.com>
	<5756A333.1010701@opensips.org>
	<CAOooq9BceX-x4qkQKXGJRyamrgqVkRKCXfQ1kNwO=LP4ZNiMPg@mail.gmail.com>
Message-ID: <578764E9.7010801@opensips.org>

Hello Ping,

Sorry for the delay - if I provide a small patch for test that will 
expose in local route and AVP holding the B2B logic ID, will be fine 
with you ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 07.06.2016 16:15, Ping Han wrote:
> Hi Bogdan,
>
> Yes, that is exactly what I am trying to do.
>
> Regards,
> Ping
>
> On Tue, Jun 7, 2016 at 8:34 PM, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Ping,
>
>     So you need the B2B Logic ID only when you create the B2B session,
>     in order to place it in the first outgoing INVITE (as extra hdr),
>     right ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 07.06.2016 04:59, Ping Han wrote:
>>     Thanks, Bogdan,
>>
>>     What I am trying to do is to get the b2b logic ID and deliver to
>>     the next hop via a custom SIP header.
>>
>>     I am using the b2b_init_request("top hiding") and it looks like
>>     the only place that I can add a custom header is in the local_route.
>>
>>     So in the script I need the b2b logic ID after
>>     running b2b_init_request("top hiding") but before the new INVITE
>>     is sent to the B party as shown below.
>>
>>     I am not sure how I can do that.
>>

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From bogdan at opensips.org  Thu Jul 14 12:14:52 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 13:14:52 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
Message-ID: <5787661C.4080604@opensips.org>

Hi Nabeel,

That means the vmusers and vmaliases do work ok, still the VM storage 
engine does not. Do you have in voicemail.conf the following:
     odbcstorage=asteriskrt
     odbctable=voicemessages


Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 05.07.2016 08:54, Nabeel wrote:
> I have been able to solve the issue of loading numbers without using 
> the voicemail.conf file.
>
> After adding the line /'voicemail => odbc,asterisk,vmaliases'/ to 
> extconfig.cfg, I removed the suffix " |u " from extensions.conf:
>
> exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)
>
>
> Now all phone numbers in the subscriber table are correctly linked to 
> their mailboxes and I can successfully retrieve voice messages.
>
> I have set 'odbcstorage=asterisk' and 'odbctable=voicemessages' in 
> voicemail.conf, but the 'voicemessages' table remains empty after 
> leaving a message.
>
> Nabeel
>
>

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From bogdan at opensips.org  Thu Jul 14 12:20:44 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 13:20:44 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
Message-ID: <5787677C.7090006@opensips.org>

Hi Nabeel,

1) that limit should not be necessary, as you should have in DB a single 
record for each subscriber. If multiple records are returned, it means 
your data is not correct.

2) in those lines, the "asterisk" and "asteriskcfg" are the names of the 
odbc connection - I pasted an example which was not correlated with the 
tutorial - just be sure you have the name like in /etc/odbc.ini and 
/etc/asterisk/res_odbc.conf

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 04.07.2016 12:54, Nabeel wrote:
>
> 0.  The line block was in the default OpenSIPS config, but I agree 
> that it is not in the tutorial so should be removed (for voicemail).
>
> 1. I think there is a misunderstanding here.  'limit' is not a column; 
> I am referring to the mysql LIMIT clause:
>
> https://dev.mysql.com/doc/refman/5.5/en/select.html
>
> 2. In the tutorial, the following is added to extconfig.cfg:
>
> sipusers => odbc,asterisk,sipusers sippeers => odbc,asterisk,sipusers 
> voicemail => odbc,asterisk,vmusers meetme => odbc,asterisk,meetme
>
> However, this is different to the config in your last Email with 
> 'asteriskcfg'. Please clarify.
>
> For the phone number +447479189410 in the subscriber table, I had to 
> add the following line to voicemail.conf for it to work (notice the " 
> |u " ):
>
> +447479189410|u => 1234,Example Mailbox,root at localhost
>
> Asterisk seems to look for the " |u " suffix in addition to the  phone 
> number, instead of just the phone number which is in the subscriber 
> table.
>
> Nabeel
>



From nabeelshikder at gmail.com  Thu Jul 14 12:33:59 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Thu, 14 Jul 2016 11:33:59 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <5787661C.4080604@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
	<5787661C.4080604@opensips.org>
Message-ID: <CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>

Hi Bogdan,

I have been able to solve that problem.
The issue was that I had asterisk compiled with file storage enabled
instead of ODBC storage. I recompiled asterisk with ODBC storage enabled
and now database storage is working.

Thanks.

Nabeel
On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org>
wrote:

> Hi Nabeel,
>
> That means the vmusers and vmaliases do work ok, still the VM storage
> engine does not. Do you have in voicemail.conf the following:
>     odbcstorage=asteriskrt
>     odbctable=voicemessages
>
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 05.07.2016 08:54, Nabeel wrote:
>
> I have been able to solve the issue of loading numbers without using the
> voicemail.conf file.
>
> After adding the line *'voicemail => odbc,asterisk,vmaliases'* to
> extconfig.cfg, I removed the suffix " |u " from extensions.conf:
>
> exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)
>
>
>
> Now all phone numbers in the subscriber table are correctly linked to
> their mailboxes and I can successfully retrieve voice messages.
>
> I have set 'odbcstorage=asterisk' and 'odbctable=voicemessages' in
> voicemail.conf, but the 'voicemessages' table remains empty after leaving a
> message.
>
> Nabeel
>
>
>
>
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From bogdan at opensips.org  Thu Jul 14 12:36:11 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 13:36:11 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
	<5787661C.4080604@opensips.org>
	<CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>
Message-ID: <57876B1B.6010409@opensips.org>

Perfect ! is there any left to be solved, or everything works fine ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 13:33, Nabeel wrote:
>
> Hi Bogdan,
>
> I have been able to solve that problem.
> The issue was that I had asterisk compiled with file storage enabled 
> instead of ODBC storage. I recompiled asterisk with ODBC storage 
> enabled and now database storage is working.
>
> Thanks.
>
> Nabeel
>
> On 14 Jul 2016 11:15 a.m., "Bogdan-Andrei Iancu" <bogdan at opensips.org 
> <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Nabeel,
>
>     That means the vmusers and vmaliases do work ok, still the VM
>     storage engine does not. Do you have in voicemail.conf the following:
>         odbcstorage=asteriskrt
>         odbctable=voicemessages
>
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>     OpenSIPS Founder and Developer
>     http://www.opensips-solutions.com
>
>     On 05.07.2016 08:54, Nabeel wrote:
>>     I have been able to solve the issue of loading numbers without
>>     using the voicemail.conf file.
>>
>>     After adding the line /'voicemail => odbc,asterisk,vmaliases'/ to
>>     extconfig.cfg, I removed the suffix " |u " from extensions.conf:
>>
>>     exten => _VMR_.,n,Voicemail(${EXTEN:4}*|u*)
>>
>>
>>     Now all phone numbers in the subscriber table are correctly
>>     linked to their mailboxes and I can successfully retrieve voice
>>     messages.
>>
>>     I have set 'odbcstorage=asterisk' and 'odbctable=voicemessages'
>>     in voicemail.conf, but the 'voicemessages' table remains empty
>>     after leaving a message.
>>
>>     Nabeel
>>
>>
>

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From nocbgtelcom at gmail.com  Thu Jul 14 13:27:19 2016
From: nocbgtelcom at gmail.com (Hristo Donev)
Date: Thu, 14 Jul 2016 14:27:19 +0300
Subject: [OpenSIPS-Users] Error in new installation
Message-ID: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>

Hello,

I try to make new installation of  OpenSips 2.2.

Here is my steps:
git clone https://github.com/OpenSIPS/opensips.git -b 2.2 opensips_2_2
cd opensips_2_2
make all
make install


if [ -z "" -a \
                        ! -f /usr/local//etc/opensips/opensips.cfg ]; then \
                mv -f /usr/local//etc/opensips/opensips.cfg.sample \
                        /usr/local//etc/opensips/opensips.cfg; \
        fi
# opensipsctl config
touch             /usr/local//etc/opensips//opensipsctlrc.sample
install -m 644 scripts/opensipsctlrc \
                /usr/local//etc/opensips//opensipsctlrc.sample
if [ ! -f /usr/local//etc/opensips//opensipsctlrc ]; then \
                mv -f /usr/local//etc/opensips//opensipsctlrc.sample \
                        /usr/local//etc/opensips//opensipsctlrc; \
        fi
# osipsconsole config
touch             /usr/local//etc/opensips//osipsconsolerc.sample
install -m 644 scripts/osipsconsolerc \
                /usr/local//etc/opensips//osipsconsolerc.sample
if [ ! -f /usr/local//etc/opensips//osipsconsolerc ]; then \
                mv -f /usr/local//etc/opensips//osipsconsolerc.sample \
                        /usr/local//etc/opensips//osipsconsolerc; \
        fi
make -C menuconfig proper
make[1]: Entering directory `/root/opensips_2_2/menuconfig'
rm -f configure
rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
make -C menuconfig \

MENUCONFIG_CFG_PATH=/usr/local//share/opensips//menuconfig_templates/ \
        MENUCONFIG_GEN_PATH=/usr/local//etc/opensips/
MENUCONFIG_HAVE_SOURCES=0
make[1]: Entering directory `/root/opensips_2_2/menuconfig'
gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops
-falign-loops -ftree-vectorize -mtune=nocona -Wold-style-definition
-Wmissing-field-initializers -Wredundant-decls -g -Wall
-DMENUCONFIG_CFG_PATH=\"/usr/local//share/opensips//menuconfig_templates/\"
-DMENUCONFIG_GEN_PATH=\"/usr/local//etc/opensips/\"
-DMENUCONFIG_HAVE_SOURCES=0    -c -o cfg.o cfg.c
In file included from main.h:31:0,
                 from cfg.c:28:
curses.h:29:19: fatal error: curses.h: No such file or directory
 #include<curses.h>
                   ^
compilation terminated.
make[1]: *** [cfg.o] Error 1
make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
make: *** [opensipsmc] Error 2


Where is problem?

Best regards,
Hristo
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From nabeelshikder at gmail.com  Thu Jul 14 13:35:24 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Thu, 14 Jul 2016 12:35:24 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <57876B1B.6010409@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
	<5787661C.4080604@opensips.org>
	<CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>
	<57876B1B.6010409@opensips.org>
Message-ID: <CA+vx6K+ArcxksfjnKNb0EHtE7xNvoxQ5Vi0YQPzB8Q-Ok1=qWg@mail.gmail.com>

I also found the correct way to deal with the LIMIT problem. Asterisk has a
built-in way to deal with this. In file* /etc/asterisk/res_odbc.conf*,  the
following should be added under [asterisk] :

limit => 5
share_connections => no

Now everything is working well without problems.

Nabeel
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From surabhi.garg at vvdntech.in  Thu Jul 14 14:17:19 2016
From: surabhi.garg at vvdntech.in (Surabhi Garg)
Date: Thu, 14 Jul 2016 17:47:19 +0530
Subject: [OpenSIPS-Users] Error in new installation
In-Reply-To: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
References: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
Message-ID: <CAAJ=DzdbNF+KRGED2C0soYBqJmr0C=U2kWeWJvgwBpmG9b0uwQ@mail.gmail.com>

Hi Hristo,

Have you installed required dependencies before doing *"make"*? Seems like
you've not.

curses.h:29:19: fatal error: curses.h: No such file or directory

The error shows that libncurses development libraries are missing.

You should install all the required dependencies before compiling OpenSIPS.

For instance on Debian:

*apt-get install libncurses5-dev*





*With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
<%2B91%209910100290>8130321886| Skype : surabhi.garg9*



On Thu, Jul 14, 2016 at 4:57 PM, Hristo Donev <nocbgtelcom at gmail.com> wrote:

> Hello,
>
> I try to make new installation of  OpenSips 2.2.
>
> Here is my steps:
> git clone https://github.com/OpenSIPS/opensips.git -b 2.2 opensips_2_2
> cd opensips_2_2
> make all
> make install
>
>
> if [ -z "" -a \
>                         ! -f /usr/local//etc/opensips/opensips.cfg ]; then
> \
>                 mv -f /usr/local//etc/opensips/opensips.cfg.sample \
>                         /usr/local//etc/opensips/opensips.cfg; \
>         fi
> # opensipsctl config
> touch             /usr/local//etc/opensips//opensipsctlrc.sample
> install -m 644 scripts/opensipsctlrc \
>                 /usr/local//etc/opensips//opensipsctlrc.sample
> if [ ! -f /usr/local//etc/opensips//opensipsctlrc ]; then \
>                 mv -f /usr/local//etc/opensips//opensipsctlrc.sample \
>                         /usr/local//etc/opensips//opensipsctlrc; \
>         fi
> # osipsconsole config
> touch             /usr/local//etc/opensips//osipsconsolerc.sample
> install -m 644 scripts/osipsconsolerc \
>                 /usr/local//etc/opensips//osipsconsolerc.sample
> if [ ! -f /usr/local//etc/opensips//osipsconsolerc ]; then \
>                 mv -f /usr/local//etc/opensips//osipsconsolerc.sample \
>                         /usr/local//etc/opensips//osipsconsolerc; \
>         fi
> make -C menuconfig proper
> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
> rm -f configure
> rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
> make -C menuconfig \
>
> MENUCONFIG_CFG_PATH=/usr/local//share/opensips//menuconfig_templates/ \
>         MENUCONFIG_GEN_PATH=/usr/local//etc/opensips/
> MENUCONFIG_HAVE_SOURCES=0
> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
> gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops
> -falign-loops -ftree-vectorize -mtune=nocona -Wold-style-definition
> -Wmissing-field-initializers -Wredundant-decls -g -Wall
> -DMENUCONFIG_CFG_PATH=\"/usr/local//share/opensips//menuconfig_templates/\"
> -DMENUCONFIG_GEN_PATH=\"/usr/local//etc/opensips/\"
> -DMENUCONFIG_HAVE_SOURCES=0    -c -o cfg.o cfg.c
> In file included from main.h:31:0,
>                  from cfg.c:28:
> curses.h:29:19: fatal error: curses.h: No such file or directory
>  #include<curses.h>
>                    ^
> compilation terminated.
> make[1]: *** [cfg.o] Error 1
> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
> make: *** [opensipsmc] Error 2
>
>
> Where is problem?
>
> Best regards,
> Hristo
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From annusfictus at gmail.com  Thu Jul 14 14:21:23 2016
From: annusfictus at gmail.com (Annus Fictus)
Date: Thu, 14 Jul 2016 14:21:23 +0200
Subject: [OpenSIPS-Users] Error in new installation
In-Reply-To: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
References: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
Message-ID: <e478e07c-47fd-d5d4-b6c0-95c038715d5b@gmail.com>

maybe ncurses-devel

on CentOS

yum install ncurses-devel

Regards

El 14/07/2016 a las 13:27, Hristo Donev escribi?:
> Hello,
>
> I try to make new installation of  OpenSips 2.2.
>
> Here is my steps:
> git clone https://github.com/OpenSIPS/opensips.git -b 2.2 opensips_2_2
> cd opensips_2_2
> make all
> make install
>
>
> if [ -z "" -a \
>                         ! -f /usr/local//etc/opensips/opensips.cfg ]; 
> then \
>                 mv -f /usr/local//etc/opensips/opensips.cfg.sample \
> /usr/local//etc/opensips/opensips.cfg; \
>         fi
> # opensipsctl config
> touch /usr/local//etc/opensips//opensipsctlrc.sample
> install -m 644 scripts/opensipsctlrc \
> /usr/local//etc/opensips//opensipsctlrc.sample
> if [ ! -f /usr/local//etc/opensips//opensipsctlrc ]; then \
>                 mv -f /usr/local//etc/opensips//opensipsctlrc.sample \
> /usr/local//etc/opensips//opensipsctlrc; \
>         fi
> # osipsconsole config
> touch /usr/local//etc/opensips//osipsconsolerc.sample
> install -m 644 scripts/osipsconsolerc \
> /usr/local//etc/opensips//osipsconsolerc.sample
> if [ ! -f /usr/local//etc/opensips//osipsconsolerc ]; then \
>                 mv -f /usr/local//etc/opensips//osipsconsolerc.sample \
> /usr/local//etc/opensips//osipsconsolerc; \
>         fi
> make -C menuconfig proper
> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
> rm -f configure
> rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
> make -C menuconfig \
> MENUCONFIG_CFG_PATH=/usr/local//share/opensips//menuconfig_templates/ \
>         MENUCONFIG_GEN_PATH=/usr/local//etc/opensips/ 
> MENUCONFIG_HAVE_SOURCES=0
> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
> gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops 
> -falign-loops -ftree-vectorize -mtune=nocona -Wold-style-definition 
> -Wmissing-field-initializers -Wredundant-decls -g -Wall 
> -DMENUCONFIG_CFG_PATH=\"/usr/local//share/opensips//menuconfig_templates/\" 
> -DMENUCONFIG_GEN_PATH=\"/usr/local//etc/opensips/\" 
> -DMENUCONFIG_HAVE_SOURCES=0    -c -o cfg.o cfg.c
> In file included from main.h:31:0,
>                  from cfg.c:28:
> curses.h:29:19: fatal error: curses.h: No such file or directory
>  #include<curses.h>
>                    ^
> compilation terminated.
> make[1]: *** [cfg.o] Error 1
> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
> make: *** [opensipsmc] Error 2
>
>
> Where is problem?
>
> Best regards,
> Hristo
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From nocbgtelcom at gmail.com  Thu Jul 14 14:57:43 2016
From: nocbgtelcom at gmail.com (Hristo Donev)
Date: Thu, 14 Jul 2016 15:57:43 +0300
Subject: [OpenSIPS-Users] Error in new installation
In-Reply-To: <CAAJ=DzdbNF+KRGED2C0soYBqJmr0C=U2kWeWJvgwBpmG9b0uwQ@mail.gmail.com>
References: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
	<CAAJ=DzdbNF+KRGED2C0soYBqJmr0C=U2kWeWJvgwBpmG9b0uwQ@mail.gmail.com>
Message-ID: <CAAf1xX-Lor0ZMtT5zkqJ8oGQ=LErzNjiafMNQk7YYBtTZrUDBw@mail.gmail.com>

For CentOs 7 no package libncurses5-dev available.


2016-07-14 15:17 GMT+03:00 Surabhi Garg <surabhi.garg at vvdntech.in>:

> Hi Hristo,
>
> Have you installed required dependencies before doing *"make"*? Seems
> like you've not.
>
> curses.h:29:19: fatal error: curses.h: No such file or directory
>
> The error shows that libncurses development libraries are missing.
>
> You should install all the required dependencies before compiling
> OpenSIPS.
>
> For instance on Debian:
>
> *apt-get install libncurses5-dev*
>
>
>
>
>
> *With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>
>
>
> On Thu, Jul 14, 2016 at 4:57 PM, Hristo Donev <nocbgtelcom at gmail.com>
> wrote:
>
>> Hello,
>>
>> I try to make new installation of  OpenSips 2.2.
>>
>> Here is my steps:
>> git clone https://github.com/OpenSIPS/opensips.git -b 2.2 opensips_2_2
>> cd opensips_2_2
>> make all
>> make install
>>
>>
>> if [ -z "" -a \
>>                         ! -f /usr/local//etc/opensips/opensips.cfg ];
>> then \
>>                 mv -f /usr/local//etc/opensips/opensips.cfg.sample \
>>                         /usr/local//etc/opensips/opensips.cfg; \
>>         fi
>> # opensipsctl config
>> touch             /usr/local//etc/opensips//opensipsctlrc.sample
>> install -m 644 scripts/opensipsctlrc \
>>                 /usr/local//etc/opensips//opensipsctlrc.sample
>> if [ ! -f /usr/local//etc/opensips//opensipsctlrc ]; then \
>>                 mv -f /usr/local//etc/opensips//opensipsctlrc.sample \
>>                         /usr/local//etc/opensips//opensipsctlrc; \
>>         fi
>> # osipsconsole config
>> touch             /usr/local//etc/opensips//osipsconsolerc.sample
>> install -m 644 scripts/osipsconsolerc \
>>                 /usr/local//etc/opensips//osipsconsolerc.sample
>> if [ ! -f /usr/local//etc/opensips//osipsconsolerc ]; then \
>>                 mv -f /usr/local//etc/opensips//osipsconsolerc.sample \
>>                         /usr/local//etc/opensips//osipsconsolerc; \
>>         fi
>> make -C menuconfig proper
>> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
>> rm -f configure
>> rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
>> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
>> make -C menuconfig \
>>
>> MENUCONFIG_CFG_PATH=/usr/local//share/opensips//menuconfig_templates/ \
>>         MENUCONFIG_GEN_PATH=/usr/local//etc/opensips/
>> MENUCONFIG_HAVE_SOURCES=0
>> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
>> gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops
>> -falign-loops -ftree-vectorize -mtune=nocona -Wold-style-definition
>> -Wmissing-field-initializers -Wredundant-decls -g -Wall
>> -DMENUCONFIG_CFG_PATH=\"/usr/local//share/opensips//menuconfig_templates/\"
>> -DMENUCONFIG_GEN_PATH=\"/usr/local//etc/opensips/\"
>> -DMENUCONFIG_HAVE_SOURCES=0    -c -o cfg.o cfg.c
>> In file included from main.h:31:0,
>>                  from cfg.c:28:
>> curses.h:29:19: fatal error: curses.h: No such file or directory
>>  #include<curses.h>
>>                    ^
>> compilation terminated.
>> make[1]: *** [cfg.o] Error 1
>> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
>> make: *** [opensipsmc] Error 2
>>
>>
>> Where is problem?
>>
>> Best regards,
>> Hristo
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From SOAN at stofa.dk  Thu Jul 14 15:02:00 2016
From: SOAN at stofa.dk (=?iso-8859-1?Q?S=F8ren_Andersen?=)
Date: Thu, 14 Jul 2016 13:02:00 +0000
Subject: [OpenSIPS-Users] drop_replies and drop_requests statistics
	counters
Message-ID: <F0A535BF32A63B4D8EC9E9AB5EC1856D72F3DCA9@SEEMBX.root.local>

Hello OpenSIPS users,

I'm starting to see opensips statistics counters increase for drop_requests and drop_replies and my b2b server is not able to make any new connections to my proxy server.
I can't figure out why my proxy start to drops the connection? - Where is no errors in my log file. Do you guys have an idea why I'm experiencing this?


/S?ren

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From nocbgtelcom at gmail.com  Thu Jul 14 15:04:08 2016
From: nocbgtelcom at gmail.com (Hristo Donev)
Date: Thu, 14 Jul 2016 16:04:08 +0300
Subject: [OpenSIPS-Users] Error in new installation
In-Reply-To: <CAAf1xX-Lor0ZMtT5zkqJ8oGQ=LErzNjiafMNQk7YYBtTZrUDBw@mail.gmail.com>
References: <CAAf1xX9OQt+ShcvscJBNBMK1rB3ta_xTTsWS9+tfTQUcAgCfkg@mail.gmail.com>
	<CAAJ=DzdbNF+KRGED2C0soYBqJmr0C=U2kWeWJvgwBpmG9b0uwQ@mail.gmail.com>
	<CAAf1xX-Lor0ZMtT5zkqJ8oGQ=LErzNjiafMNQk7YYBtTZrUDBw@mail.gmail.com>
Message-ID: <CAAf1xX9GSMXpwvmahj4Weabv1TzxH+VJ8YQB8hUcBYaiU1M+KA@mail.gmail.com>

Yes, all is OK now.
Thanks!

2016-07-14 15:57 GMT+03:00 Hristo Donev <nocbgtelcom at gmail.com>:

> For CentOs 7 no package libncurses5-dev available.
>
>
> 2016-07-14 15:17 GMT+03:00 Surabhi Garg <surabhi.garg at vvdntech.in>:
>
>> Hi Hristo,
>>
>> Have you installed required dependencies before doing *"make"*? Seems
>> like you've not.
>>
>> curses.h:29:19: fatal error: curses.h: No such file or directory
>>
>> The error shows that libncurses development libraries are missing.
>>
>> You should install all the required dependencies before compiling
>> OpenSIPS.
>>
>> For instance on Debian:
>>
>> *apt-get install libncurses5-dev*
>>
>>
>>
>>
>>
>> *With Best Regards,Surabhi GargVVDN Technologies Pvt LtdCell : +91
>> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>>
>>
>>
>> On Thu, Jul 14, 2016 at 4:57 PM, Hristo Donev <nocbgtelcom at gmail.com>
>> wrote:
>>
>>> Hello,
>>>
>>> I try to make new installation of  OpenSips 2.2.
>>>
>>> Here is my steps:
>>> git clone https://github.com/OpenSIPS/opensips.git -b 2.2 opensips_2_2
>>> cd opensips_2_2
>>> make all
>>> make install
>>>
>>>
>>> if [ -z "" -a \
>>>                         ! -f /usr/local//etc/opensips/opensips.cfg ];
>>> then \
>>>                 mv -f /usr/local//etc/opensips/opensips.cfg.sample \
>>>                         /usr/local//etc/opensips/opensips.cfg; \
>>>         fi
>>> # opensipsctl config
>>> touch             /usr/local//etc/opensips//opensipsctlrc.sample
>>> install -m 644 scripts/opensipsctlrc \
>>>                 /usr/local//etc/opensips//opensipsctlrc.sample
>>> if [ ! -f /usr/local//etc/opensips//opensipsctlrc ]; then \
>>>                 mv -f /usr/local//etc/opensips//opensipsctlrc.sample \
>>>                         /usr/local//etc/opensips//opensipsctlrc; \
>>>         fi
>>> # osipsconsole config
>>> touch             /usr/local//etc/opensips//osipsconsolerc.sample
>>> install -m 644 scripts/osipsconsolerc \
>>>                 /usr/local//etc/opensips//osipsconsolerc.sample
>>> if [ ! -f /usr/local//etc/opensips//osipsconsolerc ]; then \
>>>                 mv -f /usr/local//etc/opensips//osipsconsolerc.sample \
>>>                         /usr/local//etc/opensips//osipsconsolerc; \
>>>         fi
>>> make -C menuconfig proper
>>> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
>>> rm -f configure
>>> rm -f cfg.o curses.o items.o commands.o menus.o parser.o main.o
>>> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
>>> make -C menuconfig \
>>>
>>> MENUCONFIG_CFG_PATH=/usr/local//share/opensips//menuconfig_templates/ \
>>>         MENUCONFIG_GEN_PATH=/usr/local//etc/opensips/
>>> MENUCONFIG_HAVE_SOURCES=0
>>> make[1]: Entering directory `/root/opensips_2_2/menuconfig'
>>> gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops
>>> -falign-loops -ftree-vectorize -mtune=nocona -Wold-style-definition
>>> -Wmissing-field-initializers -Wredundant-decls -g -Wall
>>> -DMENUCONFIG_CFG_PATH=\"/usr/local//share/opensips//menuconfig_templates/\"
>>> -DMENUCONFIG_GEN_PATH=\"/usr/local//etc/opensips/\"
>>> -DMENUCONFIG_HAVE_SOURCES=0    -c -o cfg.o cfg.c
>>> In file included from main.h:31:0,
>>>                  from cfg.c:28:
>>> curses.h:29:19: fatal error: curses.h: No such file or directory
>>>  #include<curses.h>
>>>                    ^
>>> compilation terminated.
>>> make[1]: *** [cfg.o] Error 1
>>> make[1]: Leaving directory `/root/opensips_2_2/menuconfig'
>>> make: *** [opensipsmc] Error 2
>>>
>>>
>>> Where is problem?
>>>
>>> Best regards,
>>> Hristo
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
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From johan at democon.be  Thu Jul 14 15:33:33 2016
From: johan at democon.be (Johan De Clercq)
Date: Thu, 14 Jul 2016 15:33:33 +0200
Subject: [OpenSIPS-Users] rtpengine trouble.
In-Reply-To: <57875B04.6070202@opensips.org>
References: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
	<57875B04.6070202@opensips.org>
Message-ID: <CAEVWGm_zJR0yRUzzRn=7oXZkNDv5yaspmrLhPe+ptz5x-x8oEw@mail.gmail.com>

Yes indeed. That is what it seems like.

2016-07-14 11:27 GMT+02:00 Bogdan-Andrei Iancu <bogdan at opensips.org>:

> Hi Johan,
>
> So, what you say is that the rtpengine 4.0 is compatible with OpenSIPS
> 1.11 and 2.1 (using the rtpengine module ??) but not with OpenSIPS 2.2 ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 12.07.2016 09:27, Johan De Clercq wrote:
>
>
>
> Please find attached
>
> - opensips.cfg,
>
> - opensips.log (rtpengine dbg output only)
>
> - rtpengine (init.d)
>
>
> pastebin:(http://pastebin.com/ct8rCEwE)
>
> - syslog (contains opensips dbg output )
>
>
>
> Problem : opensips can?t communicate with rtpengine. (used to work with
> 1.11 and 2.1 )
>
> Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly parse UDP
> command line '2470_0 d7:command4:pinge' from 82.199.64.147:60185, using
> fallback RE
>
>
>
> Opensips is 2.2 from git; rtpengine is latest from git (4.0).
>
>
>
> Can you please indicate what my problem is or please indicate what logs
> you need ?
>
>
> Br, Johan.
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From Agalya_Ramachandran at comcast.com  Thu Jul 14 15:38:02 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 14 Jul 2016 13:38:02 +0000
Subject: [OpenSIPS-Users] Query with respect to  init script
Message-ID: <3db85bc09bac4c0182d0b8835741cd02@COPDCEX28.cable.comcast.com>

Hi team,

I have a query regarding opensips init script.
After installing opensips, right now am using "opensipsctl start/stop" to start and stop the service.
We are using centos for our project. My query is for centos we should use init script from packaging/fedora or packaging/rpm?
I have placed the init script in /etc/init.d/opensips and changed the location to /usr/local/sbin and tried to start, but it didn't.
It says" PID file /var/run/opensips.pid not readable (yet?) after start"
Which is the right command to start opensips ? Is it using opensipsctl start/stop or /etc/init.d/opensips start/stop?
And how to configure the right init script for centos and what should be done for make it working?
Please clarify me.

Regards,
Agalya

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From nick at altmann.pro  Thu Jul 14 15:54:57 2016
From: nick at altmann.pro (Nick Altmann)
Date: Thu, 14 Jul 2016 16:54:57 +0300
Subject: [OpenSIPS-Users] Query with respect to init script
In-Reply-To: <3db85bc09bac4c0182d0b8835741cd02@COPDCEX28.cable.comcast.com>
References: <3db85bc09bac4c0182d0b8835741cd02@COPDCEX28.cable.comcast.com>
Message-ID: <CAJOe1ENHiHvQU3+npkTw7MHaLFd8d6VhGHumMqSC85iRFgFe0g@mail.gmail.com>

Official packaging for centos uses specs and init-scripts from
packaging/fedora.
You may also use our official builds from http://yum.opensips.org .
If you mean centos6, start/stop commands are /etc/init.d/opensips start and
/etc/init.d/opensips stop .
If you have centos7, start/stop commands are systemctl start opensips and
systemctl stop opensips .

2016-07-14 16:38 GMT+03:00 Ramachandran, Agalya (Contractor) <
Agalya_Ramachandran at comcast.com>:

> Hi team,
>
>
>
> I have a query regarding opensips init script.
>
> After installing opensips, right now am using ?*opensipsctl start/stop?*
> to start and stop the service.
>
> We are using centos for our project. My query is for centos we should use
> init script from *packaging/fedora* or *packaging/rpm*?
>
> I have placed the init script in /etc/init.d/opensips and changed the
> location to /usr/local/sbin and tried to start, but it didn?t.
>
> It says? PID file /var/run/opensips.pid not readable (yet?) after start?
>
> Which is the right command to start opensips ? Is it using *opensipsctl
> start/stop* or */etc/init.d/opensips start/stop*?
>
> And how to configure the right init script for centos and what should be
> done for make it working?
>
> Please clarify me.
>
>
>
> Regards,
>
> Agalya
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Agalya_Ramachandran at comcast.com  Thu Jul 14 16:55:20 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 14 Jul 2016 14:55:20 +0000
Subject: [OpenSIPS-Users] Query with respect to init script
In-Reply-To: <CAJOe1ENHiHvQU3+npkTw7MHaLFd8d6VhGHumMqSC85iRFgFe0g@mail.gmail.com>
References: <3db85bc09bac4c0182d0b8835741cd02@COPDCEX28.cable.comcast.com>
	<CAJOe1ENHiHvQU3+npkTw7MHaLFd8d6VhGHumMqSC85iRFgFe0g@mail.gmail.com>
Message-ID: <496b3d1139054ee482f22569025a71c0@COPDCEX28.cable.comcast.com>

Hi Nick,

Am using centos7 and I have done the same stuffs you indicated. Copied init script from fedora and and am using systemctl start/stop. But opensips is not started. And the error is

It says? PID file /var/run/opensips.pid not readable (yet?) after start?

Any idea why PID file is not created? When I use opensipsctl start/stop it is working good and pid file is as well created.
Do we need to modify anything in init script? Would be a great help if there is some clue.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Nick Altmann
Sent: Thursday, July 14, 2016 9:55 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Query with respect to init script

Official packaging for centos uses specs and init-scripts from packaging/fedora.
You may also use our official builds from http://yum.opensips.org .
If you mean centos6, start/stop commands are /etc/init.d/opensips start and /etc/init.d/opensips stop .
If you have centos7, start/stop commands are systemctl start opensips and systemctl stop opensips .

2016-07-14 16:38 GMT+03:00 Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com<mailto:Agalya_Ramachandran at comcast.com>>:
Hi team,

I have a query regarding opensips init script.
After installing opensips, right now am using ?opensipsctl start/stop? to start and stop the service.
We are using centos for our project. My query is for centos we should use init script from packaging/fedora or packaging/rpm?
I have placed the init script in /etc/init.d/opensips and changed the location to /usr/local/sbin and tried to start, but it didn?t.
It says? PID file /var/run/opensips.pid not readable (yet?) after start?
Which is the right command to start opensips ? Is it using opensipsctl start/stop or /etc/init.d/opensips start/stop?
And how to configure the right init script for centos and what should be done for make it working?
Please clarify me.

Regards,
Agalya


_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Thu Jul 14 17:59:40 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 18:59:40 +0300
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <CA+vx6K+ArcxksfjnKNb0EHtE7xNvoxQ5Vi0YQPzB8Q-Ok1=qWg@mail.gmail.com>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
	<5787661C.4080604@opensips.org>
	<CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>
	<57876B1B.6010409@opensips.org>
	<CA+vx6K+ArcxksfjnKNb0EHtE7xNvoxQ5Vi0YQPzB8Q-Ok1=qWg@mail.gmail.com>
Message-ID: <5787B6EC.2090704@opensips.org>

Nice !

As a way of helping us (project) back, could synthesize a list with 
things that did changed since the tutorial was written ? And I will 
re-generate the tutorial, so other people will benefit from it.

Thanks and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 14:35, Nabeel wrote:
>
> I also found the correct way to deal with the LIMIT problem. Asterisk 
> has a built-in way to deal with this. In 
> file//etc/asterisk/res_odbc.conf/,  the following should be added 
> under [asterisk] :
>
> limit => 5
> share_connections => no
>
> Now everything is working well without problems.
>
> Nabeel
>

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From bogdan at opensips.org  Thu Jul 14 18:25:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 14 Jul 2016 19:25:08 +0300
Subject: [OpenSIPS-Users] opensip proxy behind firwall
In-Reply-To: <00f301d1dcec$247e61f0$6d7b25d0$@bose@rebaca.com>
References: <00fd01d1d92c$8f89dba0$ae9d92e0$@bose@rebaca.com>
	<578373D0.6020303@opensips.org>
	<00f301d1dcec$247e61f0$6d7b25d0$@bose@rebaca.com>
Message-ID: <5787BCE4.7010908@opensips.org>

Hi,

If you want the SDP to remain unaltered, why do you use the rtpproxy 
functions ??

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 13.07.2016 12:51, Priyanko Bose wrote:
>
> Hi,
>
> Thanks for your reply.
>
> I am able to make SIP call to the device (192.168.v.w(LAN) in below 
> picture) which I have registered to my opensip proxy/registrar server 
> ((LAN) 192.168.x.y), from an UAC located somewhere in the internet.
>
> But the problem is while forwarding the SDP using ?rtpproxy_offer()? 
> my proxy server rewriting the SDP with its own private IP which 
> prevents the device to find the public IP of the UAC. Hence, media is 
> not reaching the UAC.
>
> Could you please let know what are the replacement of rtpproxy_offer() 
> and rtpproxy_answer() so that the SDP remains unaltered but SIP call 
> establishes?
>
> Regards,
>
> Priyanko
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* 11 July 2016 15:54
> *To:* Priyanko Bose; users at lists.opensips.org
> *Cc:* 'Shantanu Kumar Roy'
> *Subject:* Re: opensip proxy behind firwall
>
> Hi,
>
> Could you be more specific in what the "fail to make calls" means ?
>
> Thanks,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 08.07.2016 18:22, Priyanko Bose wrote:
>
>     Hi,
>
>     I am trying to setup a OpenSIP server which should act as a SIP
>     register as well as a SIP proxy server.
>
>     If the server is placed in the same LAN where multiple SIP enabled
>     devices (e.g Cisco SX20, Polycom RPG500) reside, then the devices 
>     get registered  easily and can make SIP calls among themselves.
>
>     But everything fails when the devices are on the other side of the
>     internet and the server is behind firewall like following diagram.
>
>     ----------     ------------
>                                      -------------            
>     --------------
>
>     - Device -    <--->       - F/W   -              <---> 
>                          -  F/W    -                     <---> - 
>     OpenSIPS  -
>
>     ----------       ------------
>                                          -------------           
>      --------------
>
>     192.168.v.w(LAN)    192.168.v.1 : 63.45.x.y     (WAN)
>       220.54.w.y: 192.168.x.1       (LAN) 192.168.x.y
>
>     The server?s private IP (192.168.x.y ) is mapped to a fixed public
>     IP (220.54.w.y) and ?opensips.cfg? file has been prepared same as
>     described in the link
>     http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Server-behind-a-NAT-firewall-td5853352.html.
>     This time I am able to register the device but fails to make SIP call.
>
>     Regards,
>
>     Priyanko
>

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From feroze.chaudhry at gmail.com  Thu Jul 14 19:31:08 2016
From: feroze.chaudhry at gmail.com (feroze waris)
Date: Thu, 14 Jul 2016 22:31:08 +0500
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
	<CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>
Message-ID: <CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>

Hi Daniel and Benajmin,

Thank you for reply and sharing link of mongodb basics. Yes there is a
performance issue. I want that opensips writes and get information of
dialog, location, subscriber etc from cachedb(mongodb). For this task i
came across with the module named *db_cachedb *(
http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html ).

 According to documentation what this module do is " *The db_cachedb module
will expose the same front db api, however it will run on top of a NoSQL
back-end, emulating the SQL calls to the back-end specific queries*."

I implemented this module and also created collections in mongodb according
to mysql tables list. Now when i start opensips it gives me following error

Jul 14 17:27:45 localhost [23610]: ERROR:core:db_check_api: module
db_mongodb does not export db_use_table function
Jul 14 17:27:45 localhost [23610]: ERROR:uri:mod_init: No database module
found
Jul 14 17:27:45 localhost [23610]: ERROR:core:init_mod: failed to
initialize module uri
Jul 14 17:27:45 localhost [23610]: ERROR:core:main: error while
initializing modules

my opensips version is 2.1.3


On Wed, Jul 13, 2016 at 6:52 PM, Benjamin Cropley <
benjamin.cropley at gmail.com> wrote:

> You don't have to explicitly define a schema when you insert a document
> into MongoDB.
>
> For example, you could create a MongoDB database, and then immediately do
> an insert into an insert into a collection without even creating it or a
> 'schema'.
>
> Maybe you should do some reading about MongoDB and NoSQL in general, as a
> basic understanding should have answered that question :)
>
> https://docs.mongodb.com/manual/faq/fundamentals/
>
> On Wed, Jul 13, 2016 at 2:44 PM, feroze waris <feroze.chaudhry at gmail.com>
> wrote:
>
>> Hi,
>>
>> Can anyone tell me how to move opensips DB from SQL to NoSQL. I have seen
>> a module name db_cachedb which has a support for mongodb but how to move
>> mysql schema to mongodb schema.
>>
>> Regards
>> Feroze
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> All the best,
> Ben Cropley
> 07539 366 905
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Rahul.Gupta at ipc.com  Thu Jul 14 20:18:46 2016
From: Rahul.Gupta at ipc.com (Gupta, Rahul)
Date: Thu, 14 Jul 2016 18:18:46 +0000
Subject: [OpenSIPS-Users] Truncated Branch in Via
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577FB917.6020400@opensips.org> <5783CC6B.6040608@opensips.org>
Message-ID: <CY1PR15MB0441D88C3B6391B041FC9845F4320@CY1PR15MB0441.namprd15.prod.outlook.com>

Hi Bogdan, thanks for the prompt response and providing the fix. We have tested it and it works fine. Are you going to put this in any build ?


Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]


From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, July 11, 2016 12:42 PM
To: OpenSIPS users mailling list; Gupta, Rahul
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

There is a fix available for extra testing - see :
    https://github.com/OpenSIPS/opensips/commit/482e643469b351d12418ff54c96beee7b27dca94

please give it a try and let me know if it is solving the problem for you too.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
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From eric at uphreak.com  Thu Jul 14 20:34:49 2016
From: eric at uphreak.com (Eric Tamme)
Date: Thu, 14 Jul 2016 12:34:49 -0600
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
	<CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>
	<CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>
Message-ID: <5787DB49.5090806@uphreak.com>

Not all cachedb backends have implementations that allow you to use all 
modules.   The db api exports specific functions that may not be 
implemented for a given cache db module.  You have found an example of 
one where a module requires a db api function implementation, and the 
cachedb module you are using does not implement it.
-Eric

On 07/14/2016 11:31 AM, feroze waris wrote:
> Hi Daniel and Benajmin,
>
> Thank you for reply and sharing link of mongodb basics. Yes there is a 
> performance issue. I want that opensips writes and get information of 
> dialog, location, subscriber etc from cachedb(mongodb). For this task 
> i came across with the module named *db_cachedb *( 
> http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html ). *
>
> *
>  According to documentation what this module do is " *The db_cachedb 
> module will expose the same front db api, however it will run on top 
> of a NoSQL back-end, emulating the SQL calls to the back-end specific 
> queries*."
>
> I implemented this module and also created collections in mongodb 
> according to mysql tables list. Now when i start opensips it gives me 
> following error
>
> Jul 14 17:27:45 localhost [23610]: ERROR:core:db_check_api: module 
> db_mongodb does not export db_use_table function
> Jul 14 17:27:45 localhost [23610]: ERROR:uri:mod_init: No database 
> module found
> Jul 14 17:27:45 localhost [23610]: ERROR:core:init_mod: failed to 
> initialize module uri
> Jul 14 17:27:45 localhost [23610]: ERROR:core:main: error while 
> initializing modules
>
> my opensips version is 2.1.3
>
>
> On Wed, Jul 13, 2016 at 6:52 PM, Benjamin Cropley 
> <benjamin.cropley at gmail.com <mailto:benjamin.cropley at gmail.com>> wrote:
>
>     You don't have to explicitly define a schema when you insert a
>     document into MongoDB.
>
>     For example, you could create a MongoDB database, and then
>     immediately do an insert into an insert into a collection without
>     even creating it or a 'schema'.
>
>     Maybe you should do some reading about MongoDB and NoSQL in
>     general, as a basic understanding should have answered that
>     question :)
>
>     https://docs.mongodb.com/manual/faq/fundamentals/
>
>     On Wed, Jul 13, 2016 at 2:44 PM, feroze waris
>     <feroze.chaudhry at gmail.com <mailto:feroze.chaudhry at gmail.com>> wrote:
>
>         Hi,
>
>         Can anyone tell me how to move opensips DB from SQL to NoSQL.
>         I have seen a module name db_cachedb which has a support for
>         mongodb but how to move mysql schema to mongodb schema.
>
>         Regards
>         Feroze
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>     -- 
>     All the best,
>     Ben Cropley
>     07539 366 905
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From jim at devito.cc  Thu Jul 14 20:38:57 2016
From: jim at devito.cc (Jim DeVito)
Date: Thu, 14 Jul 2016 11:38:57 -0700
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
	<CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>
	<CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>
Message-ID: <ab37637d1036b40217059ed4aca35cfe@mail.devito.cc>

I addition to what Eric said I suspect you are failing to load 
cachedb_mongodb as stated "The following modules must be loaded before 
this module: At least one NoSQL cachedb_* module."

Thanks!!

---
Jim DeVito
Mobile 440.941.3860

On 2016-07-14 10:31, feroze waris wrote:
> Hi Daniel and Benajmin,
> 
> Thank you for reply and sharing link of mongodb basics. Yes there is a
> performance issue. I want that opensips writes and get information of
> dialog, location, subscriber etc from cachedb(mongodb). For this task
> i came across with the module named DB_CACHEDB (
> http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html [3] ).
> 
> 
>  According to documentation what this module do is " THE DB_CACHEDB
> MODULE WILL EXPOSE THE SAME FRONT DB API, HOWEVER IT WILL RUN ON TOP
> OF A NOSQL BACK-END, EMULATING THE SQL CALLS TO THE BACK-END SPECIFIC
> QUERIES."
> 
> I implemented this module and also created collections in mongodb
> according to mysql tables list. Now when i start opensips it gives me
> following error
> 
> Jul 14 17:27:45 localhost [23610]: ERROR:core:db_check_api: module
> db_mongodb does not export db_use_table function
> Jul 14 17:27:45 localhost [23610]: ERROR:uri:mod_init: No database
> module found
> Jul 14 17:27:45 localhost [23610]: ERROR:core:init_mod: failed to
> initialize module uri
> Jul 14 17:27:45 localhost [23610]: ERROR:core:main: error while
> initializing modules
> 
> my opensips version is 2.1.3
> 
> On Wed, Jul 13, 2016 at 6:52 PM, Benjamin Cropley
> <benjamin.cropley at gmail.com> wrote:
> 
>> You don't have to explicitly define a schema when you insert a
>> document into MongoDB.
>> 
>> For example, you could create a MongoDB database, and then
>> immediately do an insert into an insert into a collection without
>> even creating it or a 'schema'.
>> 
>> Maybe you should do some reading about MongoDB and NoSQL in general,
>> as a basic understanding should have answered that question :)
>> 
>> https://docs.mongodb.com/manual/faq/fundamentals/ [2]
>> 
>> On Wed, Jul 13, 2016 at 2:44 PM, feroze waris
>> <feroze.chaudhry at gmail.com> wrote:
>> 
>>> Hi,
>>> 
>>> Can anyone tell me how to move opensips DB from SQL to NoSQL. I
>>> have seen a module name db_cachedb which has a support for mongodb
>>> but how to move mysql schema to mongodb schema.
>>> 
>>> Regards
>>> Feroze
>>> 
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users [1]
>> 
>> --
>> 
>> All the best,
>> Ben Cropley
>> 07539 366 905
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users [1]
> 
> 
> 
> Links:
> ------
> [1] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> [2] https://docs.mongodb.com/manual/faq/fundamentals/
> [3] http://www.opensips.org/html/docs/modules/2.1.x/db_cachedb.html
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From Agalya_Ramachandran at comcast.com  Thu Jul 14 22:22:06 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Thu, 14 Jul 2016 20:22:06 +0000
Subject: [OpenSIPS-Users] Query with respect to init script
In-Reply-To: <496b3d1139054ee482f22569025a71c0@COPDCEX28.cable.comcast.com>
References: <3db85bc09bac4c0182d0b8835741cd02@COPDCEX28.cable.comcast.com>
	<CAJOe1ENHiHvQU3+npkTw7MHaLFd8d6VhGHumMqSC85iRFgFe0g@mail.gmail.com>
	<496b3d1139054ee482f22569025a71c0@COPDCEX28.cable.comcast.com>
Message-ID: <fff006e7515343b3a517e4bbe06de3ec@COPDCEX28.cable.comcast.com>

Nick,

I got it fixed. Opensips.default was not configured properly. Thank you.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Ramachandran, Agalya (Contractor)
Sent: Thursday, July 14, 2016 10:55 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Query with respect to init script

Hi Nick,

Am using centos7 and I have done the same stuffs you indicated. Copied init script from fedora and and am using systemctl start/stop. But opensips is not started. And the error is

It says? PID file /var/run/opensips.pid not readable (yet?) after start?

Any idea why PID file is not created? When I use opensipsctl start/stop it is working good and pid file is as well created.
Do we need to modify anything in init script? Would be a great help if there is some clue.

Regards,
Agalya

From: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> [mailto:users-bounces at lists.opensips.org] On Behalf Of Nick Altmann
Sent: Thursday, July 14, 2016 9:55 AM
To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Subject: Re: [OpenSIPS-Users] Query with respect to init script

Official packaging for centos uses specs and init-scripts from packaging/fedora.
You may also use our official builds from http://yum.opensips.org .
If you mean centos6, start/stop commands are /etc/init.d/opensips start and /etc/init.d/opensips stop .
If you have centos7, start/stop commands are systemctl start opensips and systemctl stop opensips .

2016-07-14 16:38 GMT+03:00 Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com<mailto:Agalya_Ramachandran at comcast.com>>:
Hi team,

I have a query regarding opensips init script.
After installing opensips, right now am using ?opensipsctl start/stop? to start and stop the service.
We are using centos for our project. My query is for centos we should use init script from packaging/fedora or packaging/rpm?
I have placed the init script in /etc/init.d/opensips and changed the location to /usr/local/sbin and tried to start, but it didn?t.
It says? PID file /var/run/opensips.pid not readable (yet?) after start?
Which is the right command to start opensips ? Is it using opensipsctl start/stop or /etc/init.d/opensips start/stop?
And how to configure the right init script for centos and what should be done for make it working?
Please clarify me.

Regards,
Agalya


_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From nabeelshikder at gmail.com  Fri Jul 15 18:10:13 2016
From: nabeelshikder at gmail.com (Nabeel)
Date: Fri, 15 Jul 2016 17:10:13 +0100
Subject: [OpenSIPS-Users] OpenSIPS and Asterisk integration versions
In-Reply-To: <5787B6EC.2090704@opensips.org>
References: <CA+vx6KJTOvCwAoA2ve=8yVyBUEQtQsh3YsntY-8aojY1OmLoTg@mail.gmail.com>
	<575FD7A0.6000601@opensips.org>
	<CA+vx6KKoXR+YdAMud=gVo1hapoJtvY_OJaaGouBFy4L3QVdj2g@mail.gmail.com>
	<5774E3ED.6020301@opensips.org>
	<CA+vx6KKWX-NoFcjbdnBixOsumUB4ipi59MqYmfi1eCUB89VhPw@mail.gmail.com>
	<CA+vx6K+gfC1tUuQmijhWv22YTi1Y__Ez2XtHqKoYJ9E1nXjMMA@mail.gmail.com>
	<CA+vx6K+XxvsrhD4t=qaDEhC0=5=+E3GpTQr6+4EMOUhzpG-MDQ@mail.gmail.com>
	<CA+vx6KKs_MHhxUqWsAuFx_8rN_Xe+rafnooQvvV2E7Yoo9LwDw@mail.gmail.com>
	<577A14BF.6060602@opensips.org>
	<CA+vx6K+6sZTJn3vVUAMM-3d27_B3j_+xcqHt0x53+-q0581AUg@mail.gmail.com>
	<577A28BE.1010409@opensips.org>
	<CA+vx6K+-EW-JgMhvOkRPBqa1nkDXnQHicFb8fsOoqSXtRfYWyg@mail.gmail.com>
	<CA+vx6KLvQ9h5hN3=Jv4bjK=h9iQF-Sv+sgLdbHwEtZEVBwg9nQ@mail.gmail.com>
	<5787661C.4080604@opensips.org>
	<CA+vx6KLVosc_AjyfNH8thonDioA6p+44TwVwoGH67qf-=EPzDg@mail.gmail.com>
	<57876B1B.6010409@opensips.org>
	<CA+vx6K+ArcxksfjnKNb0EHtE7xNvoxQ5Vi0YQPzB8Q-Ok1=qWg@mail.gmail.com>
	<5787B6EC.2090704@opensips.org>
Message-ID: <CA+vx6KK4+EapJy=s+V0oaS+W-naW19mvmgB-juk+s6QXvmkmXQ@mail.gmail.com>

Here is a list of changes I found:


1) Must build asterisk with ODBC storage enabled for voicemail because
using file storage will not store messages in the database.

2) Uncomment the lines *'odbcstorage=asterisk'* and
*'odbctable=voicemessages'* in voicemail.conf to enable database storage
for messages.

3) In file* /etc/asterisk/res_odbc.conf*, add the following under
[asterisk] to correctly limit database connection to the database:

limit => 5
share_connections => no
4) Change *'nat=yes'* to* 'nat=force_rport,comedia'* in the sipusers mysql
view because it is deprecated.

5) Add column 'callbackextension' to the sipusers mysql view using "*NULL
AS `callbackextension*" because it is required in the latest asterisk
version.

6) Change the priority ordering in extensions.conf to start with 1 instead
of n because the old syntax is deprecated.

7) Remove the suffix " |u " from extensions.conf because this causes
incorrect routing to mailboxes.

8) Add the line *'voicemail => odbc,asterisk,vmaliases'* to
extconfig.cfg because
this is missing in the current tutorial.

9) Remove the following parts from default opensips.cfg file because they
clash with the voicemail reply:

     if (!db_does_uri_exist()) {
>                        send_reply("420","Bad Extension");
>                        exit;
>                }
>


>                t_newtran();
>                t_reply("480", "Temporarily Unavailable");



Nabeel
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From razvan at opensips.org  Fri Jul 15 18:25:43 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 15 Jul 2016 19:25:43 +0300
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases
	schedule
Message-ID: <913bc91e-fd57-27eb-2057-af604fd4a623@opensips.org>

Hi, Everybody!

During the last weeks, along with your help, we've managed to find and 
fix a couple of bugs in the OpenSIPS 2.x series. Therefore, we decided 
to make two new releases for the concerned versions: 2.2.1 and 2.1.4.

The release is scheduled for the next Wednesday, on 20th of July 2016.

If you have any outstanding issues related to these versions, please 
report them ASAP, so we can solve them in time.

Best regards,
-- 
R?zvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com


From wasphin at gmail.com  Sat Jul 16 02:00:30 2016
From: wasphin at gmail.com (xiaofeng)
Date: Sat, 16 Jul 2016 08:00:30 +0800
Subject: [OpenSIPS-Users] opensip proxy behind firwall
In-Reply-To: <5787BCE4.7010908@opensips.org>
References: <578373D0.6020303@opensips.org>
	<5787BCE4.7010908@opensips.org>
Message-ID: <CAPE2etv_6VAOVDgXN6=XQe-KcscebKOSOFugkRmSGogn4mqj_g@mail.gmail.com>

>> I am able to make SIP call to the device (192.168.v.w(LAN) in below
picture) which I have registered to my opensip proxy/registrar server
((LAN)   192.168.x.y), from an UAC located somewhere in the internet.
>>
>> But the problem is while forwarding the SDP using ?rtpproxy_offer()? my
proxy server rewriting the SDP with its own private IP which prevents the
device to find the public IP of the UAC. Hence, media is not reaching the
UAC.
>>
>> Could you please let know what are the replacement of rtpproxy_offer()
and rtpproxy_answer() so that the SDP remains unaltered but SIP call
establishes?

In my case, I just use some function like subst_* to replace the IP with
the public IP. It works for me.
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From johan at democon.be  Sun Jul 17 09:10:44 2016
From: johan at democon.be (johan de clercq)
Date: Sun, 17 Jul 2016 09:10:44 +0200
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4
	releases	schedule
In-Reply-To: <913bc91e-fd57-27eb-2057-af604fd4a623@opensips.org>
References: <913bc91e-fd57-27eb-2057-af604fd4a623@opensips.org>
Message-ID: <011701d1dffa$579ead90$06dc08b0$@democon.be>

Hello Razvan : rtpengine 4.0 and opensips2.2 are not compatible. 
That's the only one that I am aware off. 

-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Razvan Crainea
Sent: Friday, July 15, 2016 6:26 PM
To: news at lists.opensips.org; OpenSIPS users mailling list <users at lists.opensips.org>; OpenSIPS devel mailling list <devel at lists.opensips.org>
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases schedule

Hi, Everybody!

During the last weeks, along with your help, we've managed to find and fix a couple of bugs in the OpenSIPS 2.x series. Therefore, we decided to make two new releases for the concerned versions: 2.2.1 and 2.1.4.

The release is scheduled for the next Wednesday, on 20th of July 2016.

If you have any outstanding issues related to these versions, please report them ASAP, so we can solve them in time.

Best regards,
--
R?zvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From k.galinurov at gmail.com  Sun Jul 17 13:51:48 2016
From: k.galinurov at gmail.com (Kirill Galinurov)
Date: Sun, 17 Jul 2016 14:51:48 +0300
Subject: [OpenSIPS-Users] ws wss and permissions module
Message-ID: <CAPDUut8=W4D8ZNfsPW13oYm_r3kdtxT-cR8gUp8nQa46Y3ha9g@mail.gmail.com>

  webrtc. But current version of permission module use only any?, ?udp?, ?tcp?,
    ?tls?, ?sctp?. When i try to change proto field to ws or wss i get
    :reload_address_table:
    unknown protocol field in address table, ignoring entry with id 1.
And when i try use if (check_address("0","$si","$sp", "proto")) I also
get   unknown protocol error.

    Can you fix it?


From qasimakhan at gmail.com  Sun Jul 17 19:07:05 2016
From: qasimakhan at gmail.com (qasimakhan at gmail.com)
Date: Sun, 17 Jul 2016 22:07:05 +0500
Subject: [OpenSIPS-Users] Opensips switch from SQL to NoSQL
In-Reply-To: <ab37637d1036b40217059ed4aca35cfe@mail.devito.cc>
References: <CAEUCUn=Mvw2JxZeJFTDFYxyg093x20n3Q3E0gjCOVf=Y0NgqoQ@mail.gmail.com>
	<CAF7jizKQ4fp1PELXK4=cRCfTEpde27Z3jFt3UoBP5ubsCqFpfA@mail.gmail.com>
	<CAEUCUnnMeLtbY44QSAE_ZxT4tDZWRRrDLKDnJFbpowW7HDG5gA@mail.gmail.com>
	<ab37637d1036b40217059ed4aca35cfe@mail.devito.cc>
Message-ID: <CAPBKugscbsufQWNROWdPdbC7wcn7MTiXL76m6EKSfqQd+FyAqQ@mail.gmail.com>

Dear Feroz,


From johan at democon.be  Mon Jul 18 08:48:54 2016
From: johan at democon.be (Johan De Clercq)
Date: Mon, 18 Jul 2016 08:48:54 +0200
Subject: [OpenSIPS-Users] rtpengine trouble.
In-Reply-To: <CAEVWGm_zJR0yRUzzRn=7oXZkNDv5yaspmrLhPe+ptz5x-x8oEw@mail.gmail.com>
References: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
	<57875B04.6070202@opensips.org>
	<CAEVWGm_zJR0yRUzzRn=7oXZkNDv5yaspmrLhPe+ptz5x-x8oEw@mail.gmail.com>
Message-ID: <CAEVWGm8jp6xkrvDQc2R6kEUEXgLzsSCk8iwxW+xT_erNnJHFtw@mail.gmail.com>

Yesterday I solved the issue by changing the -u parameter of rtpengine to
--listen-ng.
Now it works as it should.

Maybe it's a good idea to change the tutorial on webrtc 2.2 as the issue is
non-trivial to find.

Best regards,

2016-07-14 15:33 GMT+02:00 Johan De Clercq <johan at democon.be>:

> Yes indeed. That is what it seems like.
>
> 2016-07-14 11:27 GMT+02:00 Bogdan-Andrei Iancu <bogdan at opensips.org>:
>
>> Hi Johan,
>>
>> So, what you say is that the rtpengine 4.0 is compatible with OpenSIPS
>> 1.11 and 2.1 (using the rtpengine module ??) but not with OpenSIPS 2.2 ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 12.07.2016 09:27, Johan De Clercq wrote:
>>
>>
>>
>> Please find attached
>>
>> - opensips.cfg,
>>
>> - opensips.log (rtpengine dbg output only)
>>
>> - rtpengine (init.d)
>>
>>
>> pastebin:(http://pastebin.com/ct8rCEwE)
>>
>> - syslog (contains opensips dbg output )
>>
>>
>>
>> Problem : opensips can?t communicate with rtpengine. (used to work with
>> 1.11 and 2.1 )
>>
>> Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly parse UDP
>> command line '2470_0 d7:command4:pinge' from 82.199.64.147:60185, using
>> fallback RE
>>
>>
>>
>> Opensips is 2.2 from git; rtpengine is latest from git (4.0).
>>
>>
>>
>> Can you please indicate what my problem is or please indicate what logs
>> you need ?
>>
>>
>> Br, Johan.
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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From razvan at opensips.org  Mon Jul 18 09:26:15 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 18 Jul 2016 10:26:15 +0300
Subject: [OpenSIPS-Users] rtpengine trouble.
In-Reply-To: <CAEVWGm8jp6xkrvDQc2R6kEUEXgLzsSCk8iwxW+xT_erNnJHFtw@mail.gmail.com>
References: <CAEVWGm_BuVEG7W+saghSUJnvF3aKWQYWnozYg+S0j0d2v36nDQ@mail.gmail.com>
	<57875B04.6070202@opensips.org>
	<CAEVWGm_zJR0yRUzzRn=7oXZkNDv5yaspmrLhPe+ptz5x-x8oEw@mail.gmail.com>
	<CAEVWGm8jp6xkrvDQc2R6kEUEXgLzsSCk8iwxW+xT_erNnJHFtw@mail.gmail.com>
Message-ID: <0df32b2b-7f1b-8a09-0e67-f70f7ad9d2ea@opensips.org>

Hi, Jonah!

Thanks for debugging this. Unfortunately I've never started rtpengine 
without the -n/--listen-ng parameter, so I've never hit this issue.

However, I think the report was initially bogus, because there are no 
changes in the rtpengine module from 2.1 (in 1.11 it doesn't even 
exist), so the two versions should behave the same. This made me look 
into a different direction.

I've now added a note in the tutorial at the listening parameter[1].

[1] http://www.opensips.org/Documentation/Tutorials-WebSocket-2-2#toc3

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/18/2016 09:48 AM, Johan De Clercq wrote:
> Yesterday I solved the issue by changing the -u parameter of rtpengine 
> to --listen-ng.
> Now it works as it should.
>
> Maybe it's a good idea to change the tutorial on webrtc 2.2 as the 
> issue is non-trivial to find.
>
> Best regards,
>
> 2016-07-14 15:33 GMT+02:00 Johan De Clercq <johan at democon.be 
> <mailto:johan at democon.be>>:
>
>     Yes indeed. That is what it seems like.
>
>     2016-07-14 11:27 GMT+02:00 Bogdan-Andrei Iancu
>     <bogdan at opensips.org <mailto:bogdan at opensips.org>>:
>
>         Hi Johan,
>
>         So, what you say is that the rtpengine 4.0 is compatible with
>         OpenSIPS 1.11 and 2.1 (using the rtpengine module ??) but not
>         with OpenSIPS 2.2 ?
>
>         Regards,
>
>         Bogdan-Andrei Iancu
>         OpenSIPS Founder and Developer
>         http://www.opensips-solutions.com
>
>         On 12.07.2016 09:27, Johan De Clercq wrote:
>>
>>         Please find attached
>>
>>         - opensips.cfg,
>>
>>         - opensips.log (rtpengine dbg output only)
>>
>>         - rtpengine (init.d)
>>
>>
>>         pastebin:(http://pastebin.com/ct8rCEwE)
>>
>>         - syslog (contains opensips dbg output )
>>
>>         Problem : opensips can?t communicate with rtpengine. (used to
>>         work with 1.11 and 2.1 )
>>
>>         Jul 10 16:40:52 webrtc-1 rtpengine[2230]: Failed to properly
>>         parse UDP command line '2470_0 d7:command4:pinge' from
>>         82.199.64.147:60185 <http://82.199.64.147:60185>, using
>>         fallback RE
>>
>>         Opensips is 2.2 from git; rtpengine is latest from git (4.0).
>>
>>         Can you please indicate what my problem is or please indicate
>>         what logs you need ?
>>
>>         Br, Johan.
>>
>>
>>         _______________________________________________
>>         Users mailing list
>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From razvan at opensips.org  Mon Jul 18 09:26:51 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 18 Jul 2016 10:26:51 +0300
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases
 schedule
In-Reply-To: <011701d1dffa$579ead90$06dc08b0$@democon.be>
References: <913bc91e-fd57-27eb-2057-af604fd4a623@opensips.org>
	<011701d1dffa$579ead90$06dc08b0$@democon.be>
Message-ID: <a99065e4-5428-a681-3087-702f3f83bfe9@opensips.org>

Hi, Jonah!

This was solved in a different thread.

Thanks!

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/17/2016 10:10 AM, johan de clercq wrote:
> Hello Razvan : rtpengine 4.0 and opensips2.2 are not compatible.
> That's the only one that I am aware off.
>
> -----Original Message-----
> From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Razvan Crainea
> Sent: Friday, July 15, 2016 6:26 PM
> To: news at lists.opensips.org; OpenSIPS users mailling list <users at lists.opensips.org>; OpenSIPS devel mailling list <devel at lists.opensips.org>
> Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases schedule
>
> Hi, Everybody!
>
> During the last weeks, along with your help, we've managed to find and fix a couple of bugs in the OpenSIPS 2.x series. Therefore, we decided to make two new releases for the concerned versions: 2.2.1 and 2.1.4.
>
> The release is scheduled for the next Wednesday, on 20th of July 2016.
>
> If you have any outstanding issues related to these versions, please report them ASAP, so we can solve them in time.
>
> Best regards,
> --
> R?zvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From razvan at opensips.org  Mon Jul 18 10:11:25 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 18 Jul 2016 11:11:25 +0300
Subject: [OpenSIPS-Users] ws wss and permissions module
In-Reply-To: <CAPDUut8=W4D8ZNfsPW13oYm_r3kdtxT-cR8gUp8nQa46Y3ha9g@mail.gmail.com>
References: <CAPDUut8=W4D8ZNfsPW13oYm_r3kdtxT-cR8gUp8nQa46Y3ha9g@mail.gmail.com>
Message-ID: <ef598817-b9f9-8267-8af6-f0d895d3b616@opensips.org>

Hi, Kirill!

Thanks for reporting this! I've just pushed a fix on git. Could you 
please take the latest sources and check?

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/17/2016 02:51 PM, Kirill Galinurov wrote:
>    webrtc. But current version of permission module use only any?, ?udp?, ?tcp?,
>      ?tls?, ?sctp?. When i try to change proto field to ws or wss i get
>      :reload_address_table:
>      unknown protocol field in address table, ignoring entry with id 1.
> And when i try use if (check_address("0","$si","$sp", "proto")) I also
> get   unknown protocol error.
>
>      Can you fix it?
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From k.galinurov at gmail.com  Mon Jul 18 15:31:48 2016
From: k.galinurov at gmail.com (Kirill Galinurov)
Date: Mon, 18 Jul 2016 16:31:48 +0300
Subject: [OpenSIPS-Users] Opensips 2.2 Craches when loadbalance
Message-ID: <CAPDUut9uEgzZtzXyanagNYUwVABFn2s+HD_RVjZMuZfpdryhPw@mail.gmail.com>

Hi all. I try to load balance INVITE using load-balancer module.
My config is.
 if ((is_method("INVITE")) && (proto != WS || proto != WSS)){
               if (check_address("2","$si","$sp","any")){
                       xlog("L_INFO","+++++Incoming call from $si $sp
$proto to CallCenter\n");
                       lb_start_or_next("1","st");
                       t_relay();
                       exit;
                                         }
                       }
And opensips crashes with errrors
++++Incoming call from 192.168.75.223 5060 udp to CallCenter
Jul 18 18:10:40 [5601] DBG:load_balancer:lb_route: initial call of LB
- found requested 1/1 resource [st]
Jul 18 18:10:40 [5601] DBG:dialog:build_new_dlg: new dialog
0x7f41c59737c8 (c=903784011,f=sip:kiry at 192.168.76.237,t=sip:100 at 192.168.10.
25,ft=612508366) on hash 29
Jul 18 18:10:40 [5601] DBG:core:parse_headers: flags=ffffffffffffffff
Jul 18 18:10:40 [5601] DBG:dialog:init_leg_info: route_set , contact
sip:kiry at 192.168.76.237, cseq 20 and bind_addr udp:192.168.10.25:5
060
Jul 18 18:10:40 [5601] DBG:dialog:dlg_add_leg_info: set leg 0 for
0x7f41c59737c8: tag=<612508366> rcseq=<0>
Jul 18 18:10:40 [5601] DBG:dialog:link_dlg: ref dlg 0x7f41c59737c8
with 3 -> 3 in h_entry 0x7f41c58d6b88 - 29
Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: adding (;did=d1.2b7d0652)
Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: second RR lump found
Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: second RR lump found
Jul 18 18:10:40 [5606] DBG:core:io_wait_loop_epoll: [TCP_main]
EPOLLHUP on IN ->connection closed by the remote peer!
Jul 18 18:10:40 [5606] CRITICAL:core:receive_fd: EOF on 26
Jul 18 18:10:40 [5606] DBG:core:handle_worker: dead child 9, pid 5601
(shutting down?)
Jul 18 18:10:40 [5606] DBG:core:io_watch_del: [TCP_main] io_watch_del
op on index 14 26 (0x7e2b80, 26, 14, 0x0,0x1) fd_no=21 called
Jul 18 18:10:40 [5592] DBG:core:handle_sigs: status = 139
Jul 18 18:10:40 [5592] INFO:core:handle_sigs: child process 5601
exited by a signal 11
Jul 18 18:10:40 [5592] INFO:core:handle_sigs: core was generated
Jul 18 18:10:40 [5592] INFO:core:handle_sigs: terminating due to SIGCHLD


From bogdan at opensips.org  Mon Jul 18 15:58:18 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 18 Jul 2016 16:58:18 +0300
Subject: [OpenSIPS-Users] Opensips 2.2 Craches when loadbalance
In-Reply-To: <CAPDUut9uEgzZtzXyanagNYUwVABFn2s+HD_RVjZMuZfpdryhPw@mail.gmail.com>
References: <CAPDUut9uEgzZtzXyanagNYUwVABFn2s+HD_RVjZMuZfpdryhPw@mail.gmail.com>
Message-ID: <578CE07A.4010507@opensips.org>

Hi Kirill,

Please use the latest 2.2 sources (GIT or nightly builds for pacakges or 
tarballs) - there was a bug (in the initial 2.2 release) which was fixed 
long time ago.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 18.07.2016 16:31, Kirill Galinurov wrote:
> Hi all. I try to load balance INVITE using load-balancer module.
> My config is.
>   if ((is_method("INVITE")) && (proto != WS || proto != WSS)){
>                 if (check_address("2","$si","$sp","any")){
>                         xlog("L_INFO","+++++Incoming call from $si $sp
> $proto to CallCenter\n");
>                         lb_start_or_next("1","st");
>                         t_relay();
>                         exit;
>                                           }
>                         }
> And opensips crashes with errrors
> ++++Incoming call from 192.168.75.223 5060 udp to CallCenter
> Jul 18 18:10:40 [5601] DBG:load_balancer:lb_route: initial call of LB
> - found requested 1/1 resource [st]
> Jul 18 18:10:40 [5601] DBG:dialog:build_new_dlg: new dialog
> 0x7f41c59737c8 (c=903784011,f=sip:kiry at 192.168.76.237,t=sip:100 at 192.168.10.
> 25,ft=612508366) on hash 29
> Jul 18 18:10:40 [5601] DBG:core:parse_headers: flags=ffffffffffffffff
> Jul 18 18:10:40 [5601] DBG:dialog:init_leg_info: route_set , contact
> sip:kiry at 192.168.76.237, cseq 20 and bind_addr udp:192.168.10.25:5
> 060
> Jul 18 18:10:40 [5601] DBG:dialog:dlg_add_leg_info: set leg 0 for
> 0x7f41c59737c8: tag=<612508366> rcseq=<0>
> Jul 18 18:10:40 [5601] DBG:dialog:link_dlg: ref dlg 0x7f41c59737c8
> with 3 -> 3 in h_entry 0x7f41c58d6b88 - 29
> Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: adding (;did=d1.2b7d0652)
> Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: second RR lump found
> Jul 18 18:10:40 [5601] DBG:rr:add_rr_param: second RR lump found
> Jul 18 18:10:40 [5606] DBG:core:io_wait_loop_epoll: [TCP_main]
> EPOLLHUP on IN ->connection closed by the remote peer!
> Jul 18 18:10:40 [5606] CRITICAL:core:receive_fd: EOF on 26
> Jul 18 18:10:40 [5606] DBG:core:handle_worker: dead child 9, pid 5601
> (shutting down?)
> Jul 18 18:10:40 [5606] DBG:core:io_watch_del: [TCP_main] io_watch_del
> op on index 14 26 (0x7e2b80, 26, 14, 0x0,0x1) fd_no=21 called
> Jul 18 18:10:40 [5592] DBG:core:handle_sigs: status = 139
> Jul 18 18:10:40 [5592] INFO:core:handle_sigs: child process 5601
> exited by a signal 11
> Jul 18 18:10:40 [5592] INFO:core:handle_sigs: core was generated
> Jul 18 18:10:40 [5592] INFO:core:handle_sigs: terminating due to SIGCHLD
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From razvan at opensips.org  Wed Jul 20 12:35:59 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 20 Jul 2016 13:35:59 +0300
Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases
	schedule
In-Reply-To: <a99065e4-5428-a681-3087-702f3f83bfe9@opensips.org>
References: <913bc91e-fd57-27eb-2057-af604fd4a623@opensips.org>
	<011701d1dffa$579ead90$06dc08b0$@democon.be>
	<a99065e4-5428-a681-3087-702f3f83bfe9@opensips.org>
Message-ID: <bd382045-fde8-1ca0-efa4-01c31afa4e0b@opensips.org>

Hi, everybody!

As scheduled, two new OpenSIPS minor releases are out today: 2.2.1 and 
2.1.4!

These new releases include a series of important bug fixes, so if you 
are using OpenSIPS 2.1 or 2.2, we recommend you to upgrade your versions 
as soon as possible.

Thanks to everybody for helping us do this!

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/18/2016 10:26 AM, R?zvan Crainea wrote:
> Hi, Jonah!
>
> This was solved in a different thread.
>
> Thanks!
>
> R?zvan Crainea
> OpenSIPS Solutions
> www.opensips-solutions.com
>
> On 07/17/2016 10:10 AM, johan de clercq wrote:
>> Hello Razvan : rtpengine 4.0 and opensips2.2 are not compatible.
>> That's the only one that I am aware off.
>>
>> -----Original Message-----
>> From: users-bounces at lists.opensips.org 
>> [mailto:users-bounces at lists.opensips.org] On Behalf Of Razvan Crainea
>> Sent: Friday, July 15, 2016 6:26 PM
>> To: news at lists.opensips.org; OpenSIPS users mailling list 
>> <users at lists.opensips.org>; OpenSIPS devel mailling list 
>> <devel at lists.opensips.org>
>> Subject: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.2.1 and 2.1.4 releases 
>> schedule
>>
>> Hi, Everybody!
>>
>> During the last weeks, along with your help, we've managed to find 
>> and fix a couple of bugs in the OpenSIPS 2.x series. Therefore, we 
>> decided to make two new releases for the concerned versions: 2.2.1 
>> and 2.1.4.
>>
>> The release is scheduled for the next Wednesday, on 20th of July 2016.
>>
>> If you have any outstanding issues related to these versions, please 
>> report them ASAP, so we can solve them in time.
>>
>> Best regards,
>> -- 
>> R?zvan Crainea
>> OpenSIPS Core Developer
>> http://www.opensips-solutions.com
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From gmontassier at openip.fr  Wed Jul 20 17:47:42 2016
From: gmontassier at openip.fr (guillaume Montassier)
Date: Wed, 20 Jul 2016 17:47:42 +0200
Subject: [OpenSIPS-Users] Fraud_detection error
Message-ID: <578F9D1E.5040303@openip.fr>

Hello, everybody,

I am working on Fraud_detection on Opensips 2.1

I detected something strange:
If I set "concurrent_calls_warning" to "1" and make a call, the rc of 
check_fraud is -1 => my rule is found and warning detect, so it's OK
If I set "concurrent_calls_critical" to "1" and make a call, the rc of 
check_fraud is -2 =>  my rule is found and critical detect, so it's OK
If I set "concurrent_calls_warning" to "2" and 
"concurrent_calls_critical" to "2" and make a call, the rc of 
check_fraud is 2 =>  my rule is not found?

The result "2" seems odd to me, because the rule is found in the two 
other cases. I think I should have "1" as the result.

Best regards,

GM





From Ben.Newlin at inin.com  Fri Jul 22 15:48:28 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Fri, 22 Jul 2016 13:48:28 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
Message-ID: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>

Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com
[http://cdn.inin.com/baseimages/ININ_Logo_Signature.png]
www.inin.com<http://www.inin.com/>

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From Rahul.Gupta at ipc.com  Thu Jul 14 20:17:15 2016
From: Rahul.Gupta at ipc.com (Gupta, Rahul)
Date: Thu, 14 Jul 2016 18:17:15 +0000
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <5783CC6B.6040608@opensips.org>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577FB917.6020400@opensips.org> <5783CC6B.6040608@opensips.org>
Message-ID: <CY1PR15MB044183BA3B7DB84AA7D983EAF4320@CY1PR15MB0441.namprd15.prod.outlook.com>

Hi Bogdan, thanks for the prompt response and providing the fix. We have tested it and it works fine. Are you going to put this in any build ?


Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]


From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, July 11, 2016 12:42 PM
To: OpenSIPS users mailling list; Gupta, Rahul
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

There is a fix available for extra testing - see :
    https://github.com/OpenSIPS/opensips/commit/482e643469b351d12418ff54c96beee7b27dca94

please give it a try and let me know if it is solving the problem for you too.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 08.07.2016 17:30, Bogdan-Andrei Iancu wrote:
Hi Rahul,

Indeed, that is a good point here, thanks for bring it up. Let me investigate the code a bit and I will update you.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 07.07.2016 22:01, Gupta, Rahul wrote:
Hi Bogdan, so the problem we are facing is the endpoint has a new branch however the difference in the branch value is AFTER the MAX_BRANCH_PARAM_LEN, the highlighted text. Since opensips is copying the partial branch which is same as the previous Via and thus our SIP stack (OCCAS) is sending back 401 Unauthorized for both thinking it's a retransmit.

This is also in line with RFC3261. Shouldn't it copy the entire Via to make sure it's different from previous Message ?

REGISTER 1
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ

REGISTER 2
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915656050004_MTAuMTIuMy4xMQ


Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]



From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Thursday, July 07, 2016 4:43 AM
To: Gupta, Rahul
Cc: Elliott, Ray; users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

The received VIA hdr (with the .44 IP) is properly preserved when forwarding (in the outbound message).

The VIA hdr added by OpenSIPS ( .156 IP) is a completely new VIA and its branch value is completely independent from the branch of other VIA hdrs.

Why do they look the same ? The RFC3261 says that for stateless fwd (when basically there is no transaction, so no branch value), to avoid populating its VIA with ";branch=0" , the proxy may "copy" and use a branch value from an older VIA (a received VIA) - keep in mind it does not say to copy it entirely or part. So, OpenSIPS copies a MAX_BRANCH_PARAM_LEN length string from the previous branch param.

Everything is correct and legal (from RFC perspective).

PS: if you would use t_relay() instead of forward() - doing statefull proxy -, you will see that the branch in the VIA added by OpenSIPS will be completly different from the value in the previous VIA.

Best regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 07.07.2016 03:42, Gupta, Rahul wrote:
Hi Bogdan,

Here is a Ethernet trace (pcap) file that has a successful and an unsuccessful registration.

Frames 1-8 illustrate the successful case with Frames 2 and 6 show how opensips adds an extra VIA header that has a full VIA;branch
Frame:2
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
Via: SIP/2.0/UDP 10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8

Frames 9-16 illustrate the unsuccessful case where Frames 10 and 14 show how opensips adds an extra VIA header that has a truncated branch.
Frame 10:
Via: SIP/2.0/UDP 10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
Via: SIP/2.0/UDP 10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ

Thanks
Rahul Gupta
Systems Architect
T +1 732-690-3845  *   E rahul.gupta at ipc.com<mailto:rahul.gupta at ipc.com>

[cid:image006.jpg at 01D1940F.3E021840]



From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, July 06, 2016 5:38 AM
To: OpenSIPS users mailling list
Cc: Elliott, Ray; Gupta, Rahul
Subject: Re: [OpenSIPS-Users] Truncated Branch in Via

Hi Rahul,

That define is used to calculate the the max VIA len when OpenSIPS is generating its own VIA headers. That max len does not impact the VIA headers which were received.

I do not understand exactly (in your example) what's the flow of that VIA header. If you want, send me off-list the pcap/ngrep showing the SIP package (before and after OpenSIPS) and how it is affected.

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 05.07.2016 20:57, Gupta, Rahul wrote:
We are using opensips 1.11.5 as a stateless proxy and seeing a truncated Branch in Via for my REGISTER message. After some code digging, the MAX length is calculated using the following formula and seems like its truncating the branch after 55 characters. This is causing the REGISTER to fail in our case. Is there a config level solution to this ?

#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8 /*!<int2hex*/ + 1 /*sep*/ + \
                                MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \
                                1 /*extra space, needed by t_calc_branch*/)


truncated from opensips -->       Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
Via from EndPoint    -->                   Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ


Thanks
Rahul Gupta


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From bogdan at opensips.org  Fri Jul 15 11:25:30 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 15 Jul 2016 12:25:30 +0300
Subject: [OpenSIPS-Users] Truncated Branch in Via
In-Reply-To: <CY1PR15MB044183BA3B7DB84AA7D983EAF4320@CY1PR15MB0441.namprd15.prod.outlook.com>
References: <CY1PR15MB0441DC93DDC7E35C2830CF69F4390@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577CD180.5020802@opensips.org>
	<CY1PR15MB0441C33A53524A1F0F3FDC38F43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577E1629.6070209@opensips.org>
	<CY1PR15MB04419617F8DE47B5E19A909AF43B0@CY1PR15MB0441.namprd15.prod.outlook.com>
	<577FB917.6020400@opensips.org> <5783CC6B.6040608@opensips.org>
	<CY1PR15MB044183BA3B7DB84AA7D983EAF4320@CY1PR15MB0441.namprd15.prod.outlook.com>
Message-ID: <5788AC0A.20503@opensips.org>

Hi Rahul,

I just did the backport to all maintained versions. For the moment the 
fix will be present in all all nightly builds / rpms, until the next 
minor releases.

Thanks and regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 14.07.2016 21:17, Gupta, Rahul wrote:
>
> Hi Bogdan, thanks for the prompt response and providing the fix. We 
> have tested it and it works fine. Are you going to put this in any build ?
>
> *Thanks*
>
> *Rahul Gupta***
>
> Systems Architect
>
> *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com <mailto:rahul.gupta at ipc.com>
>
> cid:image006.jpg at 01D1940F.3E021840
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Monday, July 11, 2016 12:42 PM
> *To:* OpenSIPS users mailling list; Gupta, Rahul
> *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
> Hi Rahul,
>
> There is a fix available for extra testing - see :
> https://github.com/OpenSIPS/opensips/commit/482e643469b351d12418ff54c96beee7b27dca94
>
> please give it a try and let me know if it is solving the problem for 
> you too.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 08.07.2016 17:30, Bogdan-Andrei Iancu wrote:
>
>     Hi Rahul,
>
>     Indeed, that is a good point here, thanks for bring it up. Let me
>     investigate the code a bit and I will update you.
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 07.07.2016 22:01, Gupta, Rahul wrote:
>
>         Hi Bogdan, so the problem we are facing is the endpoint has a
>         new branch however the difference in the branch value is
>         *AFTER* the MAX_BRANCH_PARAM_LEN, the highlighted text.Since
>         opensips is copying the partial branch which is same as the
>         previous Via and thus our SIP stack (OCCAS) is sending back
>         401 Unauthorized for both thinking it?s a retransmit.
>
>         This is also in line with RFC3261. Shouldn?t it copy the
>         entire Via to make sure it?s different from previous Message ?
>
>         *REGISTER 1*
>
>         Via: SIP/2.0/UDP
>         10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
>         Via: SIP/2.0/UDP
>         10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
>         *REGISTER 2*
>
>         Via: SIP/2.0/UDP
>         10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
>         Via: SIP/2.0/UDP
>         10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915656050004_MTAuMTIuMy4xMQ
>
>         *Thanks*
>
>         *Rahul Gupta*
>
>         Systems Architect
>
>         *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com
>         <mailto:rahul.gupta at ipc.com>
>
>         cid:image006.jpg at 01D1940F.3E021840
>
>         *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>         *Sent:* Thursday, July 07, 2016 4:43 AM
>         *To:* Gupta, Rahul
>         *Cc:* Elliott, Ray; users at lists.opensips.org
>         <mailto:users at lists.opensips.org>
>         *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
>         Hi Rahul,
>
>         The received VIA hdr (with the .44 IP) is properly preserved
>         when forwarding (in the outbound message).
>
>         The VIA hdr added by OpenSIPS ( .156 IP) is a completely new
>         VIA and its branch value is completely independent from the
>         branch of other VIA hdrs.
>
>         Why do they look the same ? The RFC3261 says that for
>         stateless fwd (when basically there is no transaction, so no
>         branch value), to avoid populating its VIA with ";branch=0" ,
>         the proxy may "copy" and use a branch value from an older VIA
>         (a received VIA) - keep in mind it does not say to copy it
>         entirely or part. So, OpenSIPS copies a MAX_BRANCH_PARAM_LEN
>         length string from the previous branch param.
>
>         Everything is correct and legal (from RFC perspective).
>
>         PS: if you would use t_relay() instead of forward() - doing
>         statefull proxy -, you will see that the branch in the VIA
>         added by OpenSIPS will be completly different from the value
>         in the previous VIA.
>
>         Best regards,
>
>
>         Bogdan-Andrei Iancu
>
>         OpenSIPS Founder and Developer
>
>         http://www.opensips-solutions.com
>
>         On 07.07.2016 03:42, Gupta, Rahul wrote:
>
>             Hi Bogdan,
>
>             Here is a Ethernet trace (pcap) file that has a successful
>             and an unsuccessful registration.
>
>             Frames 1-8 illustrate the successful case with Frames 2
>             and 6 show how opensips adds an extra VIA header that has
>             a full VIA;branch
>
>             Frame:2
>
>             Via: SIP/2.0/UDP
>             10.204.70.156:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>             Via: SIP/2.0/UDP
>             10.204.45.122:5060;branch=z9hG4bK-c45d6-2ff0ce63-4583dc45-6bd144f8
>
>             Frames 9-16 illustrate the unsuccessful case where Frames
>             10 and 14 show how opensips adds an extra VIA header that
>             has a truncated branch.
>
>             Frame 10:
>
>             Via: SIP/2.0/UDP
>             10.204.70.156:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_11391
>
>             Via: SIP/2.0/UDP
>             10.205.236.44:5060;branch=z9hG4bK1c68e33e-848e-412a-9137-4fb065a7b7eb_0efbfc5e_113915620064228_MTAuMTIuMy4xMQ
>
>             *Thanks*
>
>             *Rahul Gupta*
>
>             Systems Architect
>
>             *T*+1 732-690-3845 |*E* rahul.gupta at ipc.com
>             <mailto:rahul.gupta at ipc.com>
>
>             cid:image006.jpg at 01D1940F.3E021840
>
>             *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>             *Sent:* Wednesday, July 06, 2016 5:38 AM
>             *To:* OpenSIPS users mailling list
>             *Cc:* Elliott, Ray; Gupta, Rahul
>             *Subject:* Re: [OpenSIPS-Users] Truncated Branch in Via
>
>             Hi Rahul,
>
>             That define is used to calculate the the max VIA len when
>             OpenSIPS is generating its own VIA headers. That max len
>             does not impact the VIA headers which were received.
>
>             I do not understand exactly (in your example) what's the
>             flow of that VIA header. If you want, send me off-list the
>             pcap/ngrep showing the SIP package (before and after
>             OpenSIPS) and how it is affected.
>
>             Regards,
>
>
>
>             Bogdan-Andrei Iancu
>
>             OpenSIPS Founder and Developer
>
>             http://www.opensips-solutions.com
>
>             On 05.07.2016 20:57, Gupta, Rahul wrote:
>
>                 We are using opensips 1.11.5 as a stateless proxy and
>                 seeing a truncated Branch in Via for my REGISTER
>                 message. After some code digging, the MAX length is
>                 calculated using the following formula and seems like
>                 its truncating the branch after 55 characters. This is
>                 causing the REGISTER to fail in our case. Is there a
>                 config level solution to this ?
>
>                 *#define MAX_BRANCH_PARAM_LEN  (MCOOKIE_LEN+8
>                 /*!<int2hex*/ + 1 /*sep*/ + \*
>
>                 *MD5_LEN + 1 /*!<sep*/ + 8 /*int2hex*/ + \*
>
>                 *1 /*extra space, needed by t_calc_branch*/)*
>
>                 truncated from opensips ?Via: SIP/2.0/UDP
>                 XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_11160
>
>
>                 Via from EndPoint ?Via: SIP/2.0/UDP
>                 XX.XX.XX.XX:5060;branch=z9hG4bK0fddbbc9-1487-4755-a0b3-0c319155b8c3_0efbfc5e_1116078308924346_MTAuMTIuMy4xMQ
>
>
>                 *Thanks*
>
>                 *Rahul Gupta*
>
>                 DISCLAIMER: This e-mail may contain information that
>                 is confidential, privileged or otherwise protected
>                 from disclosure. If you are not an intended recipient
>                 of this e-mail, do not duplicate or redistribute it by
>                 any means. Please delete it and any attachments and
>                 notify the sender that you have received it in error.
>                 Unintended recipients are prohibited from taking
>                 action on the basis of information in this e-mail.
>                 E-mail messages may contain computer viruses or other
>                 defects, may not be accurately replicated on other
>                 systems, or may be intercepted, deleted or interfered
>                 with without the knowledge of the sender or the
>                 intended recipient. If you are not comfortable with
>                 the risks associated with e-mail messages, you may
>                 decide not to use e-mail to communicate with IPC. IPC
>                 reserves the right, to the extent and under
>                 circumstances permitted by applicable law, to retain,
>                 monitor and intercept e-mail messages to and from its
>                 systems.
>
>
>
>
>
>
>                 _______________________________________________
>
>                 Users mailing list
>
>                 Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From spanda at 3clogic.com  Mon Jul 25 10:15:58 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 25 Jul 2016 13:45:58 +0530
Subject: [OpenSIPS-Users] Permission module use .
Message-ID: <CALf-Q44k4r-t6UzJP9M6i+u51s=putzzAEWH6Fn1an-bn3m1eA@mail.gmail.com>

Hi All ,

      I am using openisn 1.11 . And my requirment is to block all the
register requests having a perticualr ip in the contact uri .

       Basically I wanted to block looping in the server . When ever a
register request will come having the servers ip in the contact uri , I
wanted to block this .

        How can I do this ? Is permission module is needed to achieve this
or without this I can solve my problem .

     I am new to permission module so its bit confusing . Please help me if
anybody has done this earlier .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From surabhi.garg at vvdntech.in  Mon Jul 25 13:42:20 2016
From: surabhi.garg at vvdntech.in (Surabhi Garg)
Date: Mon, 25 Jul 2016 17:12:20 +0530
Subject: [OpenSIPS-Users] Permission module use .
In-Reply-To: <CALf-Q44k4r-t6UzJP9M6i+u51s=putzzAEWH6Fn1an-bn3m1eA@mail.gmail.com>
References: <CALf-Q44k4r-t6UzJP9M6i+u51s=putzzAEWH6Fn1an-bn3m1eA@mail.gmail.com>
Message-ID: <CAAJ=Dzf3w6W4RLVPES8MBehR0Unhsf0Sj+_HuE=Y4zh0jMkxiQ@mail.gmail.com>

Hi  Sasmita,

Yes, permission module could be used for this. The permission module
provides function *"allow_registe*r*()"* which can be used in the
configuration script before saving the contact in your OpenSIPS
configuration file.

Below are the steps:

   - Create a *"register.deny" *file (this contains the details of IP
   addresses you'd like to block). Also, you could create *"register.allow"
   *file (this contains the details of IP addresses which will be always
   allowed).
   - Add *allow_register() *in your OpenSIPS configuration file as shown
   below in sample.


*NOTE: Sample of register.allow & register.deny file will be present in the
source code directory like /home/xxx/opensips/module/permissions/config/
for instance. You could refer them.*
*OpenSIPS configuration snippet for blocking IP:*

 if (is_method("REGISTER"))
        {
                if
(allow_register("/usr/local/etc/opensips/register.allow",
"/usr/local/etc/opensips/register.deny")) {
                        if (!save("location"))
                                sl_reply_error();
                } else  {
                        sl_send_reply("403", "Forbidden");
                };

                exit;
        }


*IMPORTANT: I'm assuming here that you'll be getting OpenSIPS own domain
address in the Contact. As far I know if you use OpenSIPS in proxy mode it
won't provide it's address in Contact header by default. This may happen if
OpenSIPS being used in B2BUA mode. *








*With Best Regards,Surabhi GargCell : +91
<%2B91%209910100290>8130321886| Skype : surabhi.garg9*



On Mon, Jul 25, 2016 at 1:45 PM, Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi All ,
>
>       I am using openisn 1.11 . And my requirment is to block all the
> register requests having a perticualr ip in the contact uri .
>
>        Basically I wanted to block looping in the server . When ever a
> register request will come having the servers ip in the contact uri , I
> wanted to block this .
>
>         How can I do this ? Is permission module is needed to achieve this
> or without this I can solve my problem .
>
>      I am new to permission module so its bit confusing . Please help me
> if anybody has done this earlier .
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From spanda at 3clogic.com  Mon Jul 25 13:49:12 2016
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 25 Jul 2016 17:19:12 +0530
Subject: [OpenSIPS-Users] Permission module use .
In-Reply-To: <CAAJ=Dzf3w6W4RLVPES8MBehR0Unhsf0Sj+_HuE=Y4zh0jMkxiQ@mail.gmail.com>
References: <CALf-Q44k4r-t6UzJP9M6i+u51s=putzzAEWH6Fn1an-bn3m1eA@mail.gmail.com>
	<CAAJ=Dzf3w6W4RLVPES8MBehR0Unhsf0Sj+_HuE=Y4zh0jMkxiQ@mail.gmail.com>
Message-ID: <CALf-Q44CcuN9oSe6UeeYpXt+7aPYTBNvHmO7wTuaEsUinLEQ0g@mail.gmail.com>

Thanks you so much for the reply . I have done the same thing  . Finally I
got that somewhere .

In general , my opensips wont do this . But this is a special requirement
from customer to do so .


Thanks again for the info .

*Thanks & Regards*
*Sasmita Panda*
*Network Testing and Software Engineer*
*3CLogic , ph:07827611765*

On Mon, Jul 25, 2016 at 5:12 PM, Surabhi Garg <surabhi.garg at vvdntech.in>
wrote:

>
> Hi  Sasmita,
>
> Yes, permission module could be used for this. The permission module
> provides function *"allow_registe*r*()"* which can be used in the
> configuration script before saving the contact in your OpenSIPS
> configuration file.
>
> Below are the steps:
>
>    - Create a *"register.deny" *file (this contains the details of IP
>    addresses you'd like to block). Also, you could create *"register.allow"
>    *file (this contains the details of IP addresses which will be always
>    allowed).
>    - Add *allow_register() *in your OpenSIPS configuration file as shown
>    below in sample.
>
>
> *NOTE: Sample of register.allow & register.deny file will be present in
> the source code directory like
> /home/xxx/opensips/module/permissions/config/ for instance. You could refer
> them.*
> *OpenSIPS configuration snippet for blocking IP:*
>
>  if (is_method("REGISTER"))
>         {
>                 if
> (allow_register("/usr/local/etc/opensips/register.allow",
> "/usr/local/etc/opensips/register.deny")) {
>                         if (!save("location"))
>                                 sl_reply_error();
>                 } else  {
>                         sl_send_reply("403", "Forbidden");
>                 };
>
>                 exit;
>         }
>
>
> *IMPORTANT: I'm assuming here that you'll be getting OpenSIPS own domain
> address in the Contact. As far I know if you use OpenSIPS in proxy mode it
> won't provide it's address in Contact header by default. This may happen if
> OpenSIPS being used in B2BUA mode. *
>
>
>
>
>
>
>
>
> *With Best Regards,Surabhi GargCell : +91
> <%2B91%209910100290>8130321886| Skype : surabhi.garg9*
>
>
>
> On Mon, Jul 25, 2016 at 1:45 PM, Sasmita Panda <spanda at 3clogic.com> wrote:
>
>> Hi All ,
>>
>>       I am using openisn 1.11 . And my requirment is to block all the
>> register requests having a perticualr ip in the contact uri .
>>
>>        Basically I wanted to block looping in the server . When ever a
>> register request will come having the servers ip in the contact uri , I
>> wanted to block this .
>>
>>         How can I do this ? Is permission module is needed to achieve
>> this or without this I can solve my problem .
>>
>>      I am new to permission module so its bit confusing . Please help me
>> if anybody has done this earlier .
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From annusfictus at gmail.com  Mon Jul 25 17:20:08 2016
From: annusfictus at gmail.com (Annus Fictus)
Date: Mon, 25 Jul 2016 17:20:08 +0200
Subject: [OpenSIPS-Users] Save gateway used on ACC Table when using drouting
Message-ID: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>

Hello,

I'd like know if is possible save in the acc table the used gateway from 
drouting module to terminate a call.

Regards



From jim at devito.cc  Mon Jul 25 17:31:50 2016
From: jim at devito.cc (Jim DeVito)
Date: Mon, 25 Jul 2016 08:31:50 -0700
Subject: [OpenSIPS-Users] Save gateway used on ACC Table when using
 drouting
In-Reply-To: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>
References: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>
Message-ID: <86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>

Sure thing!

http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id294247

Then just add to the acc module *_extra 
"carrier_id=$avp(carrier_id_avp)" carrier_id being the column in your 
acc database or log.

Thanks!!

---
Jim DeVito


On 2016-07-25 08:20, Annus Fictus wrote:
> Hello,
> 
> I'd like know if is possible save in the acc table the used gateway
> from drouting module to terminate a call.
> 
> Regards
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From annusfictus at gmail.com  Mon Jul 25 18:01:51 2016
From: annusfictus at gmail.com (Annus Fictus)
Date: Mon, 25 Jul 2016 18:01:51 +0200
Subject: [OpenSIPS-Users] Save gateway used on ACC Table when using
 drouting
In-Reply-To: <86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>
References: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>
	<86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>
Message-ID: <47b1f266-3ff3-3545-6c9f-5bf1a208a043@gmail.com>

hello,

I tried with:

if (do_routing(,,,,"$avp(gw_id)")) {
         xlog("L_NOTICE","Gateway $avp(gw_id)");

But $avp(gw_id) is always empty.

On the module parameters:

modparam("drouting", "gw_id_avp", '$avp(gw_id)')

Any hint?

Regards


El 25/07/2016 a las 17:31, Jim DeVito escribi?:
> Sure thing!
>
> http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id294247
>
> Then just add to the acc module *_extra 
> "carrier_id=$avp(carrier_id_avp)" carrier_id being the column in your 
> acc database or log.
>
> Thanks!!
>
> ---
> Jim DeVito
>
>
> On 2016-07-25 08:20, Annus Fictus wrote:
>> Hello,
>>
>> I'd like know if is possible save in the acc table the used gateway
>> from drouting module to terminate a call.
>>
>> Regards
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From annusfictus at gmail.com  Mon Jul 25 19:24:41 2016
From: annusfictus at gmail.com (Annus Fictus)
Date: Mon, 25 Jul 2016 19:24:41 +0200
Subject: [OpenSIPS-Users] Save gateway used on ACC Table when using
 drouting [SOLVED]
In-Reply-To: <86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>
References: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>
	<86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>
Message-ID: <4ad8f608-d1eb-0c9a-40f1-c1d4894297b5@gmail.com>

hello,

this is my solution:

on the OpenSIPs configuration, ACC module:

*modparam("acc", "db_extra", "Gateway=$avp(gw_id)")*

drouting module:

*modparam("drouting", "gw_id_avp", '$avp(gw_id)')*

On opensips database:

*mysql -u root -p *

mysql> *use opensips*

mysql> *alter table acc add Gateway char(20);*

mysql> *alter table missed_calls add Gateway char(20);*

Restart OpenSIPs and try.

Regards

El 25/07/2016 a las 17:31, Jim DeVito escribi?:
> Sure thing!
>
> http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id294247
>
> Then just add to the acc module *_extra 
> "carrier_id=$avp(carrier_id_avp)" carrier_id being the column in your 
> acc database or log.
>
> Thanks!!
>
> ---
> Jim DeVito
>
>
> On 2016-07-25 08:20, Annus Fictus wrote:
>> Hello,
>>
>> I'd like know if is possible save in the acc table the used gateway
>> from drouting module to terminate a call.
>>
>> Regards
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From Agalya_Ramachandran at comcast.com  Mon Jul 25 19:25:57 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Mon, 25 Jul 2016 17:25:57 +0000
Subject: [OpenSIPS-Users] Response handling for Re-INVITE and BYE
Message-ID: <5feb314d865440cbac28d75f53b36b3b@COPDCEX28.cable.comcast.com>

Hi team,

I have a query regarding response handling for Re-Invite and Bye request.
In route relay, I have the following snippet in my config file.
route[relay] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {
                t_on_branch("per_branch_ops");
                t_on_reply("handle_nat");
                t_on_failure("missed_call");
                xlog("INFO:opensips:route[relay]:Initializing Handlers");
        }
...
}
onreply_route[handle_nat] {
...
..
xlog("INFO: incoming reply : reason is $rr; status  is $rs [from=$fu;to=$tu;rUri=$ru;callId=$ci]\n");
}

And I could able to see all the incoming response for INVITE message in onreply_route. (100,180,200OK)
But am not seeing response logged in for 200 OK for Re-INVITE and for response for BYE.
I hope all responses will hit "onreply_route" routine even after call is established.
How can I log those information? I have enabled record_route (), so till BYE for all request and responses opensips sits in between end points.
Please let me know if am missing something.

Regards,
Agalya

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From Ben.Newlin at inin.com  Mon Jul 25 19:32:02 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Mon, 25 Jul 2016 17:32:02 +0000
Subject: [OpenSIPS-Users] Response handling for Re-INVITE and BYE
Message-ID: <24A57CA5-688E-4719-A828-D1FFA3458A67@inin.com>

Failure routes, reply routes, and branch routes are all transaction specific and will only be triggered for the specific transaction for which they have been armed; in this case, the initial INVITE. If you want replies to sequential requests to also hit the reply route, you must add

t_on_reply(?handle_nat?)

in all places where you process sequential requests as well. And if you want replies to BYEs then you must include that in the method check.


Ben Newlin

From: <users-bounces at lists.opensips.org> on behalf of "Ramachandran, Agalya (Contractor)" <Agalya_Ramachandran at comcast.com>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Monday, July 25, 2016 at 1:25 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] Response handling for Re-INVITE and BYE

Hi team,

I have a query regarding response handling for Re-Invite and Bye request.
In route relay, I have the following snippet in my config file.
route[relay] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {
                t_on_branch("per_branch_ops");
                t_on_reply("handle_nat");
                t_on_failure("missed_call");
                xlog("INFO:opensips:route[relay]:Initializing Handlers");
        }
?
}
onreply_route[handle_nat] {
?
..
xlog("INFO: incoming reply : reason is $rr; status  is $rs [from=$fu;to=$tu;rUri=$ru;callId=$ci]\n");
}

And I could able to see all the incoming response for INVITE message in onreply_route. (100,180,200OK)
But am not seeing response logged in for 200 OK for Re-INVITE and for response for BYE.
I hope all responses will hit ?onreply_route? routine even after call is established.
How can I log those information? I have enabled record_route (), so till BYE for all request and responses opensips sits in between end points.
Please let me know if am missing something.

Regards,
Agalya

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From Agalya_Ramachandran at comcast.com  Mon Jul 25 20:29:47 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Mon, 25 Jul 2016 18:29:47 +0000
Subject: [OpenSIPS-Users] Response handling for Re-INVITE and BYE
In-Reply-To: <24A57CA5-688E-4719-A828-D1FFA3458A67@inin.com>
References: <24A57CA5-688E-4719-A828-D1FFA3458A67@inin.com>
Message-ID: <aabddfaf42184ea9b1f76d1534f22db1@COPDCEX28.cable.comcast.com>

Thank you Ben. I got it.

Regards,
Agalya

From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Newlin, Ben
Sent: Monday, July 25, 2016 1:32 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Response handling for Re-INVITE and BYE

Failure routes, reply routes, and branch routes are all transaction specific and will only be triggered for the specific transaction for which they have been armed; in this case, the initial INVITE. If you want replies to sequential requests to also hit the reply route, you must add

t_on_reply(?handle_nat?)

in all places where you process sequential requests as well. And if you want replies to BYEs then you must include that in the method check.


Ben Newlin

From: <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> on behalf of "Ramachandran, Agalya (Contractor)" <Agalya_Ramachandran at comcast.com<mailto:Agalya_Ramachandran at comcast.com>>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Date: Monday, July 25, 2016 at 1:25 PM
To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Subject: [OpenSIPS-Users] Response handling for Re-INVITE and BYE

Hi team,

I have a query regarding response handling for Re-Invite and Bye request.
In route relay, I have the following snippet in my config file.
route[relay] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {
                t_on_branch("per_branch_ops");
                t_on_reply("handle_nat");
                t_on_failure("missed_call");
                xlog("INFO:opensips:route[relay]:Initializing Handlers");
        }
?
}
onreply_route[handle_nat] {
?
..
xlog("INFO: incoming reply : reason is $rr; status  is $rs [from=$fu;to=$tu;rUri=$ru;callId=$ci]\n");
}

And I could able to see all the incoming response for INVITE message in onreply_route. (100,180,200OK)
But am not seeing response logged in for 200 OK for Re-INVITE and for response for BYE.
I hope all responses will hit ?onreply_route? routine even after call is established.
How can I log those information? I have enabled record_route (), so till BYE for all request and responses opensips sits in between end points.
Please let me know if am missing something.

Regards,
Agalya

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From Ben.Newlin at inin.com  Mon Jul 25 22:00:59 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Mon, 25 Jul 2016 20:00:59 +0000
Subject: [OpenSIPS-Users] How to tell if uac_auth() is successful
Message-ID: <9A19D7A7-A0A9-4D15-8586-3DC345185167@inin.com>

I am using the UAC and UAC_AUTH modules to perform trunk authentication. The uac_auth() function will only use credentials when the realm matches one in the 401/407 challenge. Is there any way to tell whether there was a successful authentication match?

The function doesn?t provide any return code (returns false even if match was found). I have tried checking for values in Authorization header ($ar, $auth.resp) but they are not populated.

If I can?t tell whether appropriate credentials were found, then I must always send another INVITE which might not have any Authorization and will just be rejected again.

Any help would be appreciated.

Thanks,
Ben Newlin
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From Agalya_Ramachandran at comcast.com  Tue Jul 26 20:26:47 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Tue, 26 Jul 2016 18:26:47 +0000
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <575E8B4E.2000708@opensips.org>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
Message-ID: <6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>

Hi Bogdan,

I have a question related to the below topic.
When am receiving more than  180 response from different destinations , am sending 1st 180 to the caller and dropping consecutive  180 's.
And this logic is working fine.

Now I have a scenario, where I need to log what are the responses sent out of Opensips.
My logic for this is, am checking the flag value. If flag is set, log the 1st 180 response, reset the flag and drop the remaining 180 response.

I could see FLAG_180 is set, and when I try to reset flag is not getting reset. Is it not possible to reset the flag values when relay is done?
I have added a route(relay2) in onreply_route  Below is the snippet of it.
Please guide me if am doing something wrong.

route[relay2] {
        if ($rs==100)
           drop();
        if ($rs==180) {
          if (isflagset(FLAG_180)){
                xlog("INFO:opensips: Flag is set \n");
                resetflag(FLAG_180);
          }
          else{
              drop();
         }
          #setflag(FLAG_180);
        }
        xlog("INFO:opensips: Sending $rs resposne out \n");
}


Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, June 13, 2016 6:31 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set.

Like:

if ($rs==180) {
    if (isflagset(FLAG_180))
        drop();
    setflag(FALG_180);
}

The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies .

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips for our project and we are currently using opensips as proxy.
Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B.
I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file?

I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller.
Can it be achieved by the changes in config file? Please guide me.

Regards,
Agalya





_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Ben.Newlin at inin.com  Tue Jul 26 22:20:32 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Tue, 26 Jul 2016 20:20:32 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
Message-ID: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>

I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin
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From bogdan at opensips.org  Wed Jul 27 09:22:24 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 27 Jul 2016 10:22:24 +0300
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
	<6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
Message-ID: <57986130.2030505@opensips.org>

Hi Agalya,

First, take care that you can set and trigger only one reply route per 
transaction!

Now, looking at your logic inthe script, it does not reflect your 
statement : keep and log the first 180, drop the others <- is this correct ?

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> I have a question related to the below topic.
>
> When am receiving more than  180 response from different destinations 
> , am sending 1^st 180 to the caller and dropping consecutive  180 ?s.
>
> And this logic is working fine.
>
> Now I have a scenario, where I need to log what are the responses sent 
> out of Opensips.
>
> My logic for this is, am checking the flag value. If flag is set, log 
> the 1^st 180 response, reset the flag and drop the remaining 180 response.
>
> I could see FLAG_180 is set, and when I try to reset flag *is not 
> getting reset*. Is it not possible to reset the flag values when relay 
> is done?
>
> I have added a route(relay2) in onreply_route  Below is the snippet of 
> it.
>
> Please guide me if am doing something wrong.
>
> route[relay2] {
>
>         if ($rs==100)
>
>            drop();
>
>         if ($rs==180) {
>
>           if (isflagset(FLAG_180)){
>
> xlog("INFO:opensips: Flag is set \n");
>
> resetflag(FLAG_180);
>
>           }
>
>           else{
>
>      drop();
>
>          }
>
> #setflag(FLAG_180);
>
>         }
>
> xlog("INFO:opensips: Sending $rs resposne out \n");
>
> }
>
> Regards,
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Monday, June 13, 2016 6:31 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; 
> Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>
> *Subject:* Re: [OpenSIPS-Users] Provisional response handling in case 
> of forking
>
> Hi Agalya,
>
> Use the onreply route (be sure to onreply_avp_mode to be set to 1 - 
> see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) 
> in combination with flags, to record when the an 180 reply was set.
>
> Like:
>
> if ($rs==180) {
>     if (isflagset(FLAG_180))
>         drop();
>     setflag(FALG_180);
> }
>
> The onreply_avp_mode 1 will ensure that the onreply route will not 
> overlap for 2 replies .
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi team,
>
>     We are using opensips for our project and we are currently using
>     opensips as proxy.
>
>     Am forking the incoming sip call, to two destinations. It Rings in
>     both Dest A and Dest B, as a result I get two 180 Ringing response
>     from A and B.
>
>     I want to filter only the first incoming 180 Ringing response and
>     send to the actual caller. Is there a way to do this in opensips
>     config file?
>
>     I have seen drop() function which drops the complete provisional
>     response. But in my case I have to forward one 180 Ringing to the
>     caller.
>
>     Can it be achieved by the changes in config file? Please guide me.
>
>     Regards,
>     Agalya
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Wed Jul 27 09:46:16 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 27 Jul 2016 10:46:16 +0300
Subject: [OpenSIPS-Users] Save gateway used on ACC Table when using
 drouting
In-Reply-To: <47b1f266-3ff3-3545-6c9f-5bf1a208a043@gmail.com>
References: <8db7c30f-97fa-4724-1278-0462fb5b271b@gmail.com>
	<86473949c8c3ff5ce3d4392429e148c2@mail.devito.cc>
	<47b1f266-3ff3-3545-6c9f-5bf1a208a043@gmail.com>
Message-ID: <579866C8.1090607@opensips.org>

Hi,

If the $avp(gw_id) is to hold the GW ID from DB table, you should not 
use it in do_routing() call - as you built that do_routing(), the 
$avp(gw_id) will be populated with the attribute field of the selected 
GW, overwriting the already set GW ID.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 25.07.2016 19:01, Annus Fictus wrote:
> hello,
>
> I tried with:
>
> if (do_routing(,,,,"$avp(gw_id)")) {
>         xlog("L_NOTICE","Gateway $avp(gw_id)");
>
> But $avp(gw_id) is always empty.
>
> On the module parameters:
>
> modparam("drouting", "gw_id_avp", '$avp(gw_id)')
>
> Any hint?
>
> Regards
>
>
> El 25/07/2016 a las 17:31, Jim DeVito escribi?:
>> Sure thing!
>>
>> http://www.opensips.org/html/docs/modules/2.2.x/drouting.html#id294247
>>
>> Then just add to the acc module *_extra 
>> "carrier_id=$avp(carrier_id_avp)" carrier_id being the column in your 
>> acc database or log.
>>
>> Thanks!!
>>
>> ---
>> Jim DeVito
>>
>>
>> On 2016-07-25 08:20, Annus Fictus wrote:
>>> Hello,
>>>
>>> I'd like know if is possible save in the acc table the used gateway
>>> from drouting module to terminate a call.
>>>
>>> Regards
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>



From bogdan at opensips.org  Wed Jul 27 09:51:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 27 Jul 2016 10:51:08 +0300
Subject: [OpenSIPS-Users] How to tell if uac_auth() is successful
In-Reply-To: <9A19D7A7-A0A9-4D15-8586-3DC345185167@inin.com>
References: <9A19D7A7-A0A9-4D15-8586-3DC345185167@inin.com>
Message-ID: <579867EC.7030609@opensips.org>

Hi Ben

The uac_auth() should return true if the authentication was successfully 
performed. Are you sure you correctly perform the test on the return 
code? Do you see any error log from the function ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 25.07.2016 23:00, Newlin, Ben wrote:
>
> I am using the UAC and UAC_AUTH modules to perform trunk 
> authentication. The uac_auth() function will only use credentials when 
> the realm matches one in the 401/407 challenge. Is there any way to 
> tell whether there was a successful authentication match?
>
> The function doesn?t provide any return code (returns false even if 
> match was found). I have tried checking for values in Authorization 
> header ($ar, $auth.resp) but they are not populated.
>
> If I can?t tell whether appropriate credentials were found, then I 
> must always send another INVITE which might not have any Authorization 
> and will just be rejected again.
>
> Any help would be appreciated.
>
> Thanks,
>
> Ben Newlin
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Jul 27 09:57:49 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 27 Jul 2016 10:57:49 +0300
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
Message-ID: <5798697D.8070407@opensips.org>

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 
SIP concepts that overlaps. You either act as an end-point (by doing 
TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions 
check and repair the routing information in the request (like Route and 
Contact headers). if you do TH, this routing info is actually hidden and 
added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you 
managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory 
debugger - see 
http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 26.07.2016 23:20, Newlin, Ben wrote:
>
> I have had 3 OpenSIPS server crashes in the last week. All were due to 
> segmentation faults. I was not able to capture core dumps; I am 
> configuring that now to catch the next crash.
>
> My logs leading up to the crash are full of errors from 
> fix_route_dialog() complaining about invalid URIs for sequential requests:
>
> Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse 
> SIP uri
>
> Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: 
> <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)
>
> Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse 
> SIP uri
>
> Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: 
> <b0i2> (4) / 
> <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)
>
> Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse 
> SIP uri
>
> Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: 
> <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)
>
> Many times the ?URI? displayed in the error message is actually 
> internal OpenSIPS variables, as in the last error above. When they are 
> from the SIP message, I have verified that the messages themselves are 
> correctly formatted. This leads me to believe there is memory 
> corruption occurring.
>
> This all started when I updated my load-balancer servers to use 
> Record-Routing, specifically the ?double_rr? mechanism for when 
> multiple interfaces exist. The Record-Routing is occurring on 
> different servers which have not crashed. Only the servers receiving 
> the Record-Routed messages are experiencing the errors.
>
> Here is a piece of the code processing sequential requests. I am using 
> the topology_hiding() functionality of the Dialog module. Are 
> validate_dialog() and fix_route_dialog() still valid in a 
> topology_hiding scenario?
>
> if (t_check_trans())
>
> setflag(SEQ_REQUEST);
>
>   if (has_totag())
>
>   {
>
> loose_route();
>
>     if (match_dialog())
>
>     {
>
>       if (!validate_dialog())
>
> fix_route_dialog();
>
>       if (is_method("BYE"))
>
> setflag(ACC_FLAG);
>
> setflag(SEQ_REQUEST);
>
>     }
>
>     else if (!isflagset(SEQ_REQUEST))
>
>     {
>
>       if (!is_method("ACK")) {
>
> route(rlog, LV_ERROR, "check_sequential", "Sequential request not 
> matched");
>
>   route(reply_error, "481", "Call Does Not Exist");
>
>       }
>
> return(EXIT);
>
>     }
>
>   }
>
> I will attempt to get core dumps of future crashes.
>
> Thanks,
>
> Ben Newlin
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Jul 27 09:59:52 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 27 Jul 2016 10:59:52 +0300
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
Message-ID: <579869F8.9060208@opensips.org>

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR 
mechanism. If you do TH, your OpenSIPS will act as and end point (from 
SIP perspective), so there will be no Route/RR headers at all. So no 
need to do loose_route or so. You just do TH matching for the sequential 
requests and nothing more.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 22.07.2016 16:48, Newlin, Ben wrote:
>
> Hi,
>
> I am using the Dialog module with topology_hiding() in my server and I 
> have a need to Record-Route the call on my server as I am advertising 
> a different address than I am listening on. I have found what I 
> believe is an inconsistency in the handling of Record-Route within the 
> Dialog topology_hiding functionality. The topology_hiding isn?t a true 
> B2BUA, but it does set up different parameters for the incoming UAC 
> and outgoing UAS sides of the call for the Via headers, Record-Route 
> and Route headers, and the Contact header(s).
>
> The problem is that the record_route() and loose_route() functions 
> operate on different sides of the call. The record_route() function 
> will only add a Record-Route header to the outgoing UAS side of the 
> call. And since the record_route() function cannot be called from 
> onreply_route, but is no way to add a Record-Route header to the UAC 
> side of the call.
>
> On the other hand, the loose_route() function only operates on the 
> incoming UAC side of the call and there is no way to perform 
> loose_route() on the UAS side of the call.
>
> So there is a situation where Record-Route headers can only be added 
> on the outgoing UAS side, but the associated Route headers can only be 
> removed on the incoming UAC side (where they won?t exist since they 
> can?t be added) and any added headers on the UAS side cannot be 
> processed properly due to the lack of loose_route.
>
> I can provide further information if this is unclear. It should be 
> easily reproducible by attempting to use record_route in a 
> topology_hiding scenario. The route is added to the outbound leg, but 
> is not removed by loose_route so the message is looped back every time.
>
> *Ben Newlin***| Sr Voice Network Engineer, PureCloud
>
> phone & fax +1.317.957.1009 | ben.newlin at inin.com
>
> www.inin.com <http://www.inin.com/>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Ben.Newlin at inin.com  Wed Jul 27 15:05:47 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Wed, 27 Jul 2016 13:05:47 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <579869F8.9060208@opensips.org>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
Message-ID: <105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>

I understand that normally you would not need RR with TH, but the two concepts are not mutually exclusive in SIP. As I said, I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. This means that TH will populate the Contact header with the advertised address and if I cannot Record-Route with the actual address then I will not receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. If you do TH, your OpenSIPS will act as and end point (from SIP perspective), so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more.

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com<mailto:ben.newlin at inin.com>
[mage removed by sender.]
www.inin.com<http://www.inin.com>





_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From pimenta at inatel.br  Wed Jul 27 15:07:44 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 13:07:44 +0000
Subject: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?
Message-ID: <BY2PR02MB16553197F11FABF1D8D2833CB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network?


--------------


P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective.


---------------


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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From achalkov at ya.ru  Wed Jul 27 15:13:03 2016
From: achalkov at ya.ru (=?utf-8?B?0KfQsNC70LrQvtCyINCQ0YDRgtGR0Lw=?=)
Date: Wed, 27 Jul 2016 16:13:03 +0300
Subject: [OpenSIPS-Users] name of event in the event message's body
Message-ID: <1003271469625183@web21h.yandex.ru>

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From Agalya_Ramachandran at comcast.com  Wed Jul 27 15:27:35 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Wed, 27 Jul 2016 13:27:35 +0000
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <57986130.2030505@opensips.org>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
	<6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
	<57986130.2030505@opensips.org>
Message-ID: <4bfa207a853f49eba59637064b77e650@COPDCEX28.cable.comcast.com>

Hi Bogdan,

Yes you are correct. Am trying to implement that logic. (keep and log the first 180, drop the others)
Am giving you snippet of onreply_route and relay2 attached here with this mail.

My intention is just to log the response sent by OpenSIPS. For other responses logic seems good and I could see that response sent out are logged.
Only for 180 resposne, am not meeting my condition.
Correct me if am wrong.

Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, July 27, 2016 3:22 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

First, take care that you can set and trigger only one reply route per transaction !

Now, looking at your logic in the script, it does not reflect your statement : keep and log the first 180, drop the others <- is this correct ?

Best regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

I have a question related to the below topic.
When am receiving more than  180 response from different destinations , am sending 1st 180 to the caller and dropping consecutive  180 's.
And this logic is working fine.

Now I have a scenario, where I need to log what are the responses sent out of Opensips.
My logic for this is, am checking the flag value. If flag is set, log the 1st 180 response, reset the flag and drop the remaining 180 response.

I could see FLAG_180 is set, and when I try to reset flag is not getting reset. Is it not possible to reset the flag values when relay is done?
I have added a route(relay2) in onreply_route  Below is the snippet of it.
Please guide me if am doing something wrong.

route[relay2] {
        if ($rs==100)
           drop();
        if ($rs==180) {
          if (isflagset(FLAG_180)){
                xlog("INFO:opensips: Flag is set \n"<INFO:opensips:Flagisset\n>);
                resetflag(FLAG_180);
          }
          else{
              drop();
         }
          #setflag(FLAG_180);
        }
        xlog("INFO:opensips: Sending $rs resposne out \n"<INFO:opensips:Sending$rsresposneout\n>);
}


Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, June 13, 2016 6:31 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set.

Like:

if ($rs==180) {
    if (isflagset(FLAG_180))
        drop();
    setflag(FALG_180);
}

The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies .

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips for our project and we are currently using opensips as proxy.
Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B.
I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file?

I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller.
Can it be achieved by the changes in config file? Please guide me.

Regards,
Agalya






_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From Ben.Newlin at inin.com  Wed Jul 27 15:44:23 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Wed, 27 Jul 2016 13:44:23 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <5798697D.8070407@opensips.org>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
Message-ID: <7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>

Bogdan,

This is a different scenario than the other you responded to. As I said, we have two types of servers that work together. One is a load-balancer and runs as a proxy. It uses double Record-Route because it sends messages between public and private networks. Then we have our other servers using TH which receive those requests. We are not using TH and RR on the same server (although I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should not be called when using TH, I believe the documentation should reference that. It states that match_dialog() must be used with TH, but does not indicate that the other functions should not be used or that the functionality won?t work. There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), validate_dialog(), and fix_route_dialog() from my script. The crash still occurred and the BT still pointed to fix_route_dialog() function. So it must be getting called from within Dialog module somewhere. That BT is here: http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. You either act as an end-point (by doing TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and repair the routing information in the request (like Route and Contact headers). if you do TH, this routing info is actually hidden and added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin




_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From johan at democon.be  Wed Jul 27 16:57:11 2016
From: johan at democon.be (Johan De Clercq)
Date: Wed, 27 Jul 2016 16:57:11 +0200
Subject: [OpenSIPS-Users] OpenSIPS is behind a NAT,
	not the UAC. What to do?
In-Reply-To: <BY2PR02MB16553197F11FABF1D8D2833CB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB16553197F11FABF1D8D2833CB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <CAEVWGm847kub6-+gSRK7Ok5tvKDcbLqY=MZ6mgaLErOwYu8EJw@mail.gmail.com>

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br>:

> Dear OpenSIPS-users.
>
>
>
> The function nat_uac_test() (from module NATHELPER) works very well and
> tells me whether a cliente to my OpenSIPS is behind a NAT.
>
>
> But, for my specific network topology, is my OpenSIPS that is behind a
> NAT, from the client perspective. In this case I have to fix the SDP
> content that goes from OpenSIPS to the client, so that the client will be
> able to send its media to a public IP, when communicating to a peer in the
> same network domain of this SIP server.
>
>
> How can I be sure that for a client perspective the OpenSIPS is behind a
> NAT? In another words, is there a way to the OpenSIPS determine if its
> client is in the same network or in a remote network?
>
>
> --------------
>
>
> P.S. I suspect that OpenSIPS should be used in a node with public IP to
> simplify our solution. But our customer asked us to put OpenSIPS in a
> residential device that will be always behind a NAT for some smartphones
> perspective and not behind a NAT for another home devices perspective.
>
>
>
> ---------------
>
>
> Any hint will be very helpful!
>
>
> Best regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From Ben.Newlin at inin.com  Wed Jul 27 18:02:44 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Wed, 27 Jul 2016 16:02:44 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
Message-ID: <204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>

I have identified that these crashes are occurring when the far end system is not returning the Record-Route headers in the 200 OK response. The headers are present in the 180 response, but not the 200 OK. I have reproduced the scenario using SIPp and captured a SIP trace: http://pastebin.com/ckKk3EhY

The crash occurs on receipt of the ACK request and attempt to match the dialog.

I also captured a BT for this scenario as well, in case anything specific in the trace made the issue easier to find: http://pastebin.com/cM3FhPiw

I am working with the other system to try to fix their behavior.

Ideally the Record-Route headers from previous replies could be used in this case to allow the call to succeed, but I don?t know if that is possible.

Thanks,

Ben Newlin

From: "Newlin, Ben" <Ben.Newlin at inin.com>
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-Andrei Iancu <bogdan at opensips.org>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Bogdan,

This is a different scenario than the other you responded to. As I said, we have two types of servers that work together. One is a load-balancer and runs as a proxy. It uses double Record-Route because it sends messages between public and private networks. Then we have our other servers using TH which receive those requests. We are not using TH and RR on the same server (although I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should not be called when using TH, I believe the documentation should reference that. It states that match_dialog() must be used with TH, but does not indicate that the other functions should not be used or that the functionality won?t work. There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), validate_dialog(), and fix_route_dialog() from my script. The crash still occurred and the BT still pointed to fix_route_dialog() function. So it must be getting called from within Dialog module somewhere. That BT is here: http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. You either act as an end-point (by doing TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and repair the routing information in the request (like Route and Contact headers). if you do TH, this routing info is actually hidden and added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin





_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From pimenta at inatel.br  Wed Jul 27 19:04:36 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 17:04:36 +0000
Subject: [OpenSIPS-Users] OpenSIPS is behind a NAT,
 not the UAC. What to do?
In-Reply-To: <CAEVWGm847kub6-+gSRK7Ok5tvKDcbLqY=MZ6mgaLErOwYu8EJw@mail.gmail.com>
References: <BY2PR02MB16553197F11FABF1D8D2833CB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>,
	<CAEVWGm847kub6-+gSRK7Ok5tvKDcbLqY=MZ6mgaLErOwYu8EJw@mail.gmail.com>
Message-ID: <BY2PR02MB165524381012F8BBD9ABE5ACB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


It sounds good for who is using rtpproxy. In our case we are using direct media without rtpproxy.


As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have decided to do the following:






RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Johan De Clercq <johan at democon.be>
Enviado: quarta-feira, 27 de julho de 2016 11:57
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br<mailto:pimenta at inatel.br>>:

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network?


--------------


P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective.


---------------


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From pimenta at inatel.br  Wed Jul 27 19:08:34 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 17:08:34 +0000
Subject: [OpenSIPS-Users] OpenSIPS is behind a NAT,
 not the UAC. What to do?
In-Reply-To: <BY2PR02MB165524381012F8BBD9ABE5ACB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB16553197F11FABF1D8D2833CB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>,
	<CAEVWGm847kub6-+gSRK7Ok5tvKDcbLqY=MZ6mgaLErOwYu8EJw@mail.gmail.com>,
	<BY2PR02MB165524381012F8BBD9ABE5ACB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <BY2PR02MB16559AD8F0197BB36E470D61B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Sorry. The rest of message:


1 - check the IP from SDP.

2 - If such IP is equal to the IP of the gateway, then:

        we will fix the current IP, changing it to the OpenSIPS public IP.


Hopefully, it is easy to read such IP from SDP, using some function of OpenSIPS.


Could you comment about this decision, please?


Thank you very much!



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: Rodrigo Pimenta Carvalho
Enviado: quarta-feira, 27 de julho de 2016 14:04
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?


Hi.


It sounds good for who is using rtpproxy. In our case we are using direct media without rtpproxy.


As long as our OpenSIPS will be behind a NAT and such NAT is a gateway, we have decided to do the following:






RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Johan De Clercq <johan at democon.be>
Enviado: quarta-feira, 27 de julho de 2016 11:57
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] OpenSIPS is behind a NAT, not the UAC. What to do?

in the startup route, you should find  your pub ip using curl.
Put that in a var and then use that var in rtpproxy.

2016-07-27 15:07 GMT+02:00 Rodrigo Pimenta Carvalho <pimenta at inatel.br<mailto:pimenta at inatel.br>>:

Dear OpenSIPS-users.



The function nat_uac_test() (from module NATHELPER) works very well and tells me whether a cliente to my OpenSIPS is behind a NAT.


But, for my specific network topology, is my OpenSIPS that is behind a NAT, from the client perspective. In this case I have to fix the SDP content that goes from OpenSIPS to the client, so that the client will be able to send its media to a public IP, when communicating to a peer in the same network domain of this SIP server.


How can I be sure that for a client perspective the OpenSIPS is behind a NAT? In another words, is there a way to the OpenSIPS determine if its client is in the same network or in a remote network?


--------------


P.S. I suspect that OpenSIPS should be used in a node with public IP to simplify our solution. But our customer asked us to put OpenSIPS in a residential device that will be always behind a NAT for some smartphones perspective and not behind a NAT for another home devices perspective.


---------------


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From crf at transtelco.net  Wed Jul 27 19:29:00 2016
From: crf at transtelco.net (Cesar Alberto Rodriguez Fierro)
Date: Wed, 27 Jul 2016 11:29:00 -0600
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
Message-ID: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>

Hi !

I am currently working in a project related with display in real time the
active calls of our VoIP traffic, I would like to get the active sip-calls
from a Kamailio Sip Server (running opensips), is there any way to obtain
this information.

Best Regards.




[image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112
 |

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of
the individual or entity to which it is addressed and may contain
information that is privileged, confidential, and exempt from disclosure
under applicable law.  If you are not the intended recipient of this
information, you are notified that any use, dissemination, distribution, or
copying of the communication is strictly prohibited.
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From pimenta at inatel.br  Wed Jul 27 19:34:27 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 17:34:27 +0000
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
Message-ID: <BY2PR02MB1655EB4F2B186E6457051168B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


Table dialog from the OpenSIPS database will contain information useful for your needing.

You will have to query the database to get data from such table and then handle data as you need.

For querying the database, see about http://www.opensips.org/html/docs/modules/2.2.x/avpops.html


Regards.

AVPops Module - OpenSIPS<http://www.opensips.org/html/docs/modules/2.2.x/avpops.html>
www.opensips.org
AVPops (AVP-operations) modules implements a set of script functions which allow access and manipulation of user AVPs (preferences) and pseudo-variables.




RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users at lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law.  If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited.
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From daniel.zanutti at gmail.com  Wed Jul 27 19:35:19 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Wed, 27 Jul 2016 14:35:19 -0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
Message-ID: <CAN0jgiNJO4Qu1B7FbfFRQYENsEc8WjMXCN6xLqLQ+5ZuGLo-Fg@mail.gmail.com>

Hi Cesar

For realtime querying, you can use dialog profiles. Set a profile (for
example: inbound) on all calls like this:
set_dlg_profile("inbound");

Then read this using FIFO.

I just didn't get the "Kamailio Sip Server running Opensips".

Regards

On Wed, Jul 27, 2016 at 2:29 PM, Cesar Alberto Rodriguez Fierro <
crf at transtelco.net> wrote:

> Hi !
>
> I am currently working in a project related with display in real time the
> active calls of our VoIP traffic, I would like to get the active sip-calls
> from a Kamailio Sip Server (running opensips), is there any way to obtain
> this information.
>
> Best Regards.
>
>
>
>
> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
> 656-257-4112 |
>
> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
> of the individual or entity to which it is addressed and may contain
> information that is privileged, confidential, and exempt from disclosure
> under applicable law.  If you are not the intended recipient of this
> information, you are notified that any use, dissemination, distribution, or
> copying of the communication is strictly prohibited.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From pimenta at inatel.br  Wed Jul 27 19:39:44 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 17:39:44 +0000
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
Message-ID: <BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful.


Regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users at lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law.  If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited.
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From pimenta at inatel.br  Wed Jul 27 19:42:56 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 17:42:56 +0000
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>,
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Now, thinking more about it, I would suggest you to put a SQL query in your proprietary software to query the OpenSIPS database directly.

The table dialog will be always updated about current calls.


Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br>
Enviado: quarta-feira, 27 de julho de 2016 14:39
Para: users at lists.opensips.org
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.


With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful.


Regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users at lists.opensips.org
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law.  If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited.
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From crf at transtelco.net  Wed Jul 27 20:41:50 2016
From: crf at transtelco.net (Cesar Alberto Rodriguez Fierro)
Date: Wed, 27 Jul 2016 12:41:50 -0600
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>

Thanks for your help.

I trying to use  FIFO in order to  send requests to OpenSIPS, I have read
some documentation about the  Dialog Module.  I guess using the "opensipsctl
fifo dlg_list" command can be useful to obtain the current calls, but I am
not sure why the command is not available in my OpenSIPS version.   When I
execute the command ./opensipsctl fifo version, I am getting the following
information
Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).



On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
pimenta at inatel.br> wrote:

> Now, thinking more about it, I would suggest you to put a SQL query in
> your proprietary software to query the OpenSIPS database directly.
>
> The table dialog will be always updated about current calls.
>
>
> Regards.
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> ------------------------------
> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br>
> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
> *Para:* users at lists.opensips.org
> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>
>
> With FIFO you can send requests to OpenSIPS, for example from a
> proprietary software. So, if a request wants to execute a query with
> avpops, I guess FIFO will be useful.
>
>
> Regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> ------------------------------
> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
> *Para:* users at lists.opensips.org
> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>
> Hi !
>
> I am currently working in a project related with display in real time the
> active calls of our VoIP traffic, I would like to get the active sip-calls
> from a Kamailio Sip Server (running opensips), is there any way to obtain
> this information.
>
> Best Regards.
>
>
>
>
> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
> 656-257-4112 |
>
> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
> of the individual or entity to which it is addressed and may contain
> information that is privileged, confidential, and exempt from disclosure
> under applicable law.  If you are not the intended recipient of this
> information, you are notified that any use, dissemination, distribution, or
> copying of the communication is strictly prohibited.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From daniel.zanutti at gmail.com  Wed Jul 27 21:12:24 2016
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Wed, 27 Jul 2016 16:12:24 -0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
Message-ID: <CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>

On my sample, you should run:

opensipsctl fifo profile_get_size inbound


Using dlg_list you should get something like this:
# opensipsctl fifo dlg_list
dialog::  hash=84:1411689852
        state:: 4
        user_flags:: 0
        timestart:: 1469646635
        datestart:: 2016-07-27 16:10:35
        timeout:: 1469653835
        dateout:: 2016-07-27 18:10:35
...
dialog::  hash=289:324429409
        state:: 2
        user_flags:: 0
        timestart:: 0
        timeout:: 0
...
dialog::  hash=640:1695114669
        state:: 4
...

Check if modules are successfully loaded.

Regards

On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
crf at transtelco.net> wrote:

>
> Thanks for your help.
>
> I trying to use  FIFO in order to  send requests to OpenSIPS, I have read
> some documentation about the  Dialog Module.  I guess using the "opensipsctl
> fifo dlg_list" command can be useful to obtain the current calls, but I am
> not sure why the command is not available in my OpenSIPS version.   When I
> execute the command ./opensipsctl fifo version, I am getting the following
> information
> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>
>
>
> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
> pimenta at inatel.br> wrote:
>
>> Now, thinking more about it, I would suggest you to put a SQL query in
>> your proprietary software to query the OpenSIPS database directly.
>>
>> The table dialog will be always updated about current calls.
>>
>>
>> Regards.
>>
>>
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>>
>>
>> ------------------------------
>> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
>> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br>
>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>> *Para:* users at lists.opensips.org
>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>
>>
>> With FIFO you can send requests to OpenSIPS, for example from a
>> proprietary software. So, if a request wants to execute a query with
>> avpops, I guess FIFO will be useful.
>>
>>
>> Regards.
>>
>>
>>
>> RODRIGO PIMENTA CARVALHO
>> Inatel Competence Center
>> Software
>> Ph: +55 35 3471 9200 RAMAL 979
>>
>>
>> ------------------------------
>> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
>> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>> *Para:* users at lists.opensips.org
>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>
>> Hi !
>>
>> I am currently working in a project related with display in real time the
>> active calls of our VoIP traffic, I would like to get the active sip-calls
>> from a Kamailio Sip Server (running opensips), is there any way to obtain
>> this information.
>>
>> Best Regards.
>>
>>
>>
>>
>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>> 656-257-4112 |
>>
>> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
>> of the individual or entity to which it is addressed and may contain
>> information that is privileged, confidential, and exempt from disclosure
>> under applicable law.  If you are not the intended recipient of this
>> information, you are notified that any use, dissemination, distribution, or
>> copying of the communication is strictly prohibited.
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From kaduww at gmail.com  Wed Jul 27 21:20:17 2016
From: kaduww at gmail.com (Carlos Eduardo)
Date: Wed, 27 Jul 2016 16:20:17 -0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
Message-ID: <CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>

Cesar,

Are you using the dialog module in your script?

This MI command will only return a valid value if the dialog module is
loaded (loadmodule "dialog.so") and if the dialogs are criated durign the
script processing (create_dialog() command).

2016-07-27 16:12 GMT-03:00 Daniel Zanutti <daniel.zanutti at gmail.com>:

> On my sample, you should run:
>
> opensipsctl fifo profile_get_size inbound
>
>
> Using dlg_list you should get something like this:
> # opensipsctl fifo dlg_list
> dialog::  hash=84:1411689852
>         state:: 4
>         user_flags:: 0
>         timestart:: 1469646635
>         datestart:: 2016-07-27 16:10:35
>         timeout:: 1469653835
>         dateout:: 2016-07-27 18:10:35
> ...
> dialog::  hash=289:324429409
>         state:: 2
>         user_flags:: 0
>         timestart:: 0
>         timeout:: 0
> ...
> dialog::  hash=640:1695114669
>         state:: 4
> ...
>
> Check if modules are successfully loaded.
>
> Regards
>
> On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
> crf at transtelco.net> wrote:
>
>>
>> Thanks for your help.
>>
>> I trying to use  FIFO in order to  send requests to OpenSIPS, I have
>> read some documentation about the  Dialog Module.  I guess using the "opensipsctl
>> fifo dlg_list" command can be useful to obtain the current calls, but I am
>> not sure why the command is not available in my OpenSIPS version.   When I
>> execute the command ./opensipsctl fifo version, I am getting the following
>> information
>> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>>
>>
>>
>> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
>> pimenta at inatel.br> wrote:
>>
>>> Now, thinking more about it, I would suggest you to put a SQL query in
>>> your proprietary software to query the OpenSIPS database directly.
>>>
>>> The table dialog will be always updated about current calls.
>>>
>>>
>>> Regards.
>>>
>>>
>>> RODRIGO PIMENTA CARVALHO
>>> Inatel Competence Center
>>> Software
>>> Ph: +55 35 3471 9200 RAMAL 979
>>>
>>>
>>> ------------------------------
>>> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
>>> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br>
>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>>> *Para:* users at lists.opensips.org
>>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>>
>>>
>>> With FIFO you can send requests to OpenSIPS, for example from a
>>> proprietary software. So, if a request wants to execute a query with
>>> avpops, I guess FIFO will be useful.
>>>
>>>
>>> Regards.
>>>
>>>
>>>
>>> RODRIGO PIMENTA CARVALHO
>>> Inatel Competence Center
>>> Software
>>> Ph: +55 35 3471 9200 RAMAL 979
>>>
>>>
>>> ------------------------------
>>> *De:* users-bounces at lists.opensips.org <users-bounces at lists.opensips.org>
>>> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net>
>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>>> *Para:* users at lists.opensips.org
>>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>>
>>> Hi !
>>>
>>> I am currently working in a project related with display in real time
>>> the active calls of our VoIP traffic, I would like to get the active
>>> sip-calls from a Kamailio Sip Server (running opensips), is there any way
>>> to obtain this information.
>>>
>>> Best Regards.
>>>
>>>
>>>
>>>
>>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>>> 656-257-4112 |
>>>
>>> CONFIDENTIALITY NOTICE:  This communication is intended only for the use
>>> of the individual or entity to which it is addressed and may contain
>>> information that is privileged, confidential, and exempt from disclosure
>>> under applicable law.  If you are not the intended recipient of this
>>> information, you are notified that any use, dissemination, distribution, or
>>> copying of the communication is strictly prohibited.
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
*Carlos E. Wagner*
*Tecn?logo em Telecomunica??es, OCP, dCAA*

*Gnotel Telecom*
*E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
            *carlos at gnotel.com.br <carlos at gnotel.com.br>*
*Fone:* +55 48 9981-0894
*Skype:* carlos.e.wagner
www.gnotel.com.br
www.blogdovoip.com
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From pimenta at inatel.br  Wed Jul 27 21:48:34 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 19:48:34 +0000
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>,
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
Message-ID: <BY2PR02MB1655762C9BC81237CCEBDCABB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


In my case I used to run:


echo $'dlg_list\n' | xargs ./opensipsctl fifo > RespostasFIFO.txt               or                     ./opensipsctl fifo dlg_list > RespostasFIFO.txt


The file RespostasFIFO.txt will be created automatically.

I my script I also have loadmodule "dialog.so".


Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979


________________________________
De: users-bounces at lists.opensips.org <users-bounces at lists.opensips.org> em nome de Carlos Eduardo <kaduww at gmail.com>
Enviado: quarta-feira, 27 de julho de 2016 16:20
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.

Cesar,

Are you using the dialog module in your script?

This MI command will only return a valid value if the dialog module is loaded (loadmodule "dialog.so") and if the dialogs are criated durign the script processing (create_dialog() command).

2016-07-27 16:12 GMT-03:00 Daniel Zanutti <daniel.zanutti at gmail.com<mailto:daniel.zanutti at gmail.com>>:
On my sample, you should run:

opensipsctl fifo profile_get_size inbound


Using dlg_list you should get something like this:
# opensipsctl fifo dlg_list
dialog::  hash=84:1411689852
        state:: 4
        user_flags:: 0
        timestart:: 1469646635
        datestart:: 2016-07-27 16:10:35
        timeout:: 1469653835
        dateout:: 2016-07-27 18:10:35
...
dialog::  hash=289:324429409
        state:: 2
        user_flags:: 0
        timestart:: 0
        timeout:: 0
...
dialog::  hash=640:1695114669
        state:: 4
...

Check if modules are successfully loaded.

Regards

On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <crf at transtelco.net<mailto:crf at transtelco.net>> wrote:

Thanks for your help.

I trying to use  FIFO in order to  send requests to OpenSIPS, I have read some documentation about the  Dialog Module.  I guess using the "opensipsctl fifo dlg_list" command can be useful to obtain the current calls, but I am not sure why the command is not available in my OpenSIPS version.   When I execute the command ./opensipsctl fifo version, I am getting the following information
Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).



On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <pimenta at inatel.br<mailto:pimenta at inatel.br>> wrote:

Now, thinking more about it, I would suggest you to put a SQL query in your proprietary software to query the OpenSIPS database directly.

The table dialog will be always updated about current calls.


Regards.


RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979


________________________________
De: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> em nome de Rodrigo Pimenta Carvalho <pimenta at inatel.br<mailto:pimenta at inatel.br>>
Enviado: quarta-feira, 27 de julho de 2016 14:39
Para: users at lists.opensips.org<mailto:users at lists.opensips.org>
Assunto: Re: [OpenSIPS-Users] Get concurrent calls from sip server.


With FIFO you can send requests to OpenSIPS, for example from a proprietary software. So, if a request wants to execute a query with avpops, I guess FIFO will be useful.


Regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979


________________________________
De: users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org> <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> em nome de Cesar Alberto Rodriguez Fierro <crf at transtelco.net<mailto:crf at transtelco.net>>
Enviado: quarta-feira, 27 de julho de 2016 14:29
Para: users at lists.opensips.org<mailto:users at lists.opensips.org>
Assunto: [OpenSIPS-Users] Get concurrent calls from sip server.

Hi !

I am currently working in a project related with display in real time the active calls of our VoIP traffic, I would like to get the active sip-calls from a Kamailio Sip Server (running opensips), is there any way to obtain this information.

Best Regards.




[Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52 656-257-4112<tel:%2B52%20656-257-4112> |
[https://ssl.gstatic.com/ui/v1/icons/mail/images/cleardot.gif]

CONFIDENTIALITY NOTICE:  This communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential, and exempt from disclosure under applicable law.  If you are not the intended recipient of this information, you are notified that any use, dissemination, distribution, or copying of the communication is strictly prohibited.

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users



_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users




--
Carlos E. Wagner
Tecn?logo em Telecomunica??es, OCP, dCAA

Gnotel Telecom
E-mail: kaduww at gmail.com<mailto:kaduww at gmail.com>
            carlos at gnotel.com.br<mailto:carlos at gnotel.com.br>
Fone: +55 48 9981-0894
Skype: carlos.e.wagner
www.gnotel.com.br<http://www.gnotel.com.br>
www.blogdovoip.com<http://www.blogdovoip.com>
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From crf at transtelco.net  Wed Jul 27 21:57:11 2016
From: crf at transtelco.net (Cesar Alberto Rodriguez Fierro)
Date: Wed, 27 Jul 2016 13:57:11 -0600
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
Message-ID: <CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>

There was a wrong configuration of the  "dialog.so" module.

I will try to parse the output in order to obtain the current calls.

 Thanks for the help.

On Wed, Jul 27, 2016 at 1:20 PM, Carlos Eduardo <kaduww at gmail.com> wrote:

> Cesar,
>
> Are you using the dialog module in your script?
>
> This MI command will only return a valid value if the dialog module is
> loaded (loadmodule "dialog.so") and if the dialogs are criated durign the
> script processing (create_dialog() command).
>
> 2016-07-27 16:12 GMT-03:00 Daniel Zanutti <daniel.zanutti at gmail.com>:
>
>> On my sample, you should run:
>>
>> opensipsctl fifo profile_get_size inbound
>>
>>
>> Using dlg_list you should get something like this:
>> # opensipsctl fifo dlg_list
>> dialog::  hash=84:1411689852
>>         state:: 4
>>         user_flags:: 0
>>         timestart:: 1469646635
>>         datestart:: 2016-07-27 16:10:35
>>         timeout:: 1469653835
>>         dateout:: 2016-07-27 18:10:35
>> ...
>> dialog::  hash=289:324429409
>>         state:: 2
>>         user_flags:: 0
>>         timestart:: 0
>>         timeout:: 0
>> ...
>> dialog::  hash=640:1695114669
>>         state:: 4
>> ...
>>
>> Check if modules are successfully loaded.
>>
>> Regards
>>
>> On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
>> crf at transtelco.net> wrote:
>>
>>>
>>> Thanks for your help.
>>>
>>> I trying to use  FIFO in order to  send requests to OpenSIPS, I have
>>> read some documentation about the  Dialog Module.  I guess using the "opensipsctl
>>> fifo dlg_list" command can be useful to obtain the current calls, but I am
>>> not sure why the command is not available in my OpenSIPS version.   When I
>>> execute the command ./opensipsctl fifo version, I am getting the following
>>> information
>>> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>>>
>>>
>>>
>>> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
>>> pimenta at inatel.br> wrote:
>>>
>>>> Now, thinking more about it, I would suggest you to put a SQL query in
>>>> your proprietary software to query the OpenSIPS database directly.
>>>>
>>>> The table dialog will be always updated about current calls.
>>>>
>>>>
>>>> Regards.
>>>>
>>>>
>>>> RODRIGO PIMENTA CARVALHO
>>>> Inatel Competence Center
>>>> Software
>>>> Ph: +55 35 3471 9200 RAMAL 979
>>>>
>>>>
>>>> ------------------------------
>>>> *De:* users-bounces at lists.opensips.org <
>>>> users-bounces at lists.opensips.org> em nome de Rodrigo Pimenta Carvalho <
>>>> pimenta at inatel.br>
>>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>>>> *Para:* users at lists.opensips.org
>>>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>>>
>>>>
>>>> With FIFO you can send requests to OpenSIPS, for example from a
>>>> proprietary software. So, if a request wants to execute a query with
>>>> avpops, I guess FIFO will be useful.
>>>>
>>>>
>>>> Regards.
>>>>
>>>>
>>>>
>>>> RODRIGO PIMENTA CARVALHO
>>>> Inatel Competence Center
>>>> Software
>>>> Ph: +55 35 3471 9200 RAMAL 979
>>>>
>>>>
>>>> ------------------------------
>>>> *De:* users-bounces at lists.opensips.org <
>>>> users-bounces at lists.opensips.org> em nome de Cesar Alberto Rodriguez
>>>> Fierro <crf at transtelco.net>
>>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>>>> *Para:* users at lists.opensips.org
>>>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>>>
>>>> Hi !
>>>>
>>>> I am currently working in a project related with display in real time
>>>> the active calls of our VoIP traffic, I would like to get the active
>>>> sip-calls from a Kamailio Sip Server (running opensips), is there any way
>>>> to obtain this information.
>>>>
>>>> Best Regards.
>>>>
>>>>
>>>>
>>>>
>>>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>>>> 656-257-4112 |
>>>>
>>>> CONFIDENTIALITY NOTICE:  This communication is intended only for the
>>>> use of the individual or entity to which it is addressed and may contain
>>>> information that is privileged, confidential, and exempt from disclosure
>>>> under applicable law.  If you are not the intended recipient of this
>>>> information, you are notified that any use, dissemination, distribution, or
>>>> copying of the communication is strictly prohibited.
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> --
> *Carlos E. Wagner*
> *Tecn?logo em Telecomunica??es, OCP, dCAA*
>
> *Gnotel Telecom*
> *E-mail:* *kaduww at gmail.com <kaduww at gmail.com>*
>             *carlos at gnotel.com.br <carlos at gnotel.com.br>*
> *Fone:* +55 48 9981-0894
> *Skype:* carlos.e.wagner
> www.gnotel.com.br
> www.blogdovoip.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From pimenta at inatel.br  Wed Jul 27 22:14:49 2016
From: pimenta at inatel.br (Rodrigo Pimenta Carvalho)
Date: Wed, 27 Jul 2016 20:14:49 +0000
Subject: [OpenSIPS-Users] Is it possible to read parts of SDP? What is the
	module to do it?
Message-ID: <BY2PR02MB16556BE602A333909C33C2DEB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>

Hi.


A SDP message is:


v=0
o=Z 0 0 IN IP4 192.168.21.40
s=Z
c=IN IP4 192.168.21.40
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv


It can be in a SIP INVITE or in a SIP OK message.


How can I read the IP4 from there, in case of SIP INVITE or SIP OK, and get the value 192.168.21.40 for example?

Is there a module and function that provides such information in my script?


Any hint will be very helpful!


Best regards.



RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
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From bogdan at opensips.org  Wed Jul 27 23:51:02 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 28 Jul 2016 00:51:02 +0300
Subject: [OpenSIPS-Users] [OpenSIPS-News] OpenSIPS and FreeSWITCH
	Integration Training
In-Reply-To: <57640ACB.3020101@opensips.org>
References: <57640ACB.3020101@opensips.org>
Message-ID: <57992CC6.2020707@opensips.org>

Hello,

The detailed description of the training content can be found here :
http://www.opensips.org/pub/events/2016-08-12_OpenSIPS-ClueCon_Chicago/ClueConTraining.pdf

See you in Chicago,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 17.06.2016 17:35, Bogdan-Andrei Iancu wrote:
> Hello,
>
> As part of ClueCon, we will run an one-day hands-on Training around 
> OpenSIPS and FreeSWITCH integration. The training follows the steps of 
> building a complete system featuring:
>     * OpenSIPS as a cluster front-end  - http://opensips.org
>     * FreeSWITCH PBX core system  - http://freeswitch.org/
>     * HOMER for SIP capturing  - http://sipcapture.org/
>     * CGRates as billing engine  - http://cgrates.org/
>     * SIP Fraud detection
>
> The training will be held on 12th of August 2016, as part of the 
> ClueCon event.
>
> Together with the training, we will run a new session of Design 
> Clinics (http://www.opensips.org/Community/Clinics)
>
> Details, registration (both training and clinics) and more are 
> available here:
>     http://www.opensips.org/events/Training-2016ClueCon.html
>
>
> Best regards
>



From bogdan at opensips.org  Wed Jul 27 23:58:08 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 28 Jul 2016 00:58:08 +0300
Subject: [OpenSIPS-Users] Is it possible to read parts of SDP? What is
 the module to do it?
In-Reply-To: <BY2PR02MB16556BE602A333909C33C2DEB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
References: <BY2PR02MB16556BE602A333909C33C2DEB50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
Message-ID: <57992E70.7090901@opensips.org>

Hi Rodrigo,

Take a look at the {sdp} transformation, it might help you:
     http://www.opensips.org/Documentation/Script-Tran-2-2#toc79

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 27.07.2016 23:14, Rodrigo Pimenta Carvalho wrote:
>
> Hi.
>
>
> A SDP message is:
>
>
> v=0
> o=Z 0 0 IN IP4 192.168.21.40
> s=Z
> c=IN IP4 192.168.21.40
> t=0 0
> m=audio 8000 RTP/AVP 3 110 8 0 98 101
> a=rtpmap:110 speex/8000
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
>
> It can be in a SIP INVITE or in a SIP OK message.
>
>
> How can I read the IP4 from there, in case of SIP INVITE or SIP OK, 
> and get the value 192.168.21.40 for example?
>
> Is there a module and function that provides such information in my 
> script?
>
>
> Any hint will be very helpful!
>
>
> Best regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From millennium.bug at gmail.com  Thu Jul 28 06:41:29 2016
From: millennium.bug at gmail.com (Owais Ahmad)
Date: Thu, 28 Jul 2016 09:41:29 +0500
Subject: [OpenSIPS-Users] [OpenSIPS-News] OpenSIPS and FreeSWITCH
 Integration Training
In-Reply-To: <57992CC6.2020707@opensips.org>
References: <57640ACB.3020101@opensips.org> <57992CC6.2020707@opensips.org>
Message-ID: <CAArAGKkugLJGp9OizQgLL7b4zTg7fMjJAAq0ZVdHdjv+eReZfQ@mail.gmail.com>

Hi Bogdan,

Will this training session be webcast?

Regards,
Owais

On Thu, Jul 28, 2016 at 2:51 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hello,
>
> The detailed description of the training content can be found here :
>
> http://www.opensips.org/pub/events/2016-08-12_OpenSIPS-ClueCon_Chicago/ClueConTraining.pdf
>
> See you in Chicago,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 17.06.2016 17:35, Bogdan-Andrei Iancu wrote:
>
>> Hello,
>>
>> As part of ClueCon, we will run an one-day hands-on Training around
>> OpenSIPS and FreeSWITCH integration. The training follows the steps of
>> building a complete system featuring:
>>     * OpenSIPS as a cluster front-end  - http://opensips.org
>>     * FreeSWITCH PBX core system  - http://freeswitch.org/
>>     * HOMER for SIP capturing  - http://sipcapture.org/
>>     * CGRates as billing engine  - http://cgrates.org/
>>     * SIP Fraud detection
>>
>> The training will be held on 12th of August 2016, as part of the ClueCon
>> event.
>>
>> Together with the training, we will run a new session of Design Clinics (
>> http://www.opensips.org/Community/Clinics)
>>
>> Details, registration (both training and clinics) and more are available
>> here:
>>     http://www.opensips.org/events/Training-2016ClueCon.html
>>
>>
>> Best regards
>>
>>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From gmaruzz at gmail.com  Thu Jul 28 11:45:06 2016
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 28 Jul 2016 11:45:06 +0200
Subject: [OpenSIPS-Users] [OpenSIPS-News] OpenSIPS and FreeSWITCH
 Integration Training
In-Reply-To: <CAArAGKkugLJGp9OizQgLL7b4zTg7fMjJAAq0ZVdHdjv+eReZfQ@mail.gmail.com>
References: <57640ACB.3020101@opensips.org> <57992CC6.2020707@opensips.org>
	<CAArAGKkugLJGp9OizQgLL7b4zTg7fMjJAAq0ZVdHdjv+eReZfQ@mail.gmail.com>
Message-ID: <CALXCt0p2yyM1bFaJhsXeS1+wRj92JAZrryDis4VXmj9nGn45ig@mail.gmail.com>

No, it's a for pay training seminar, you must attend in person.

Other OpenSIPS trainings are available, to be attended remotely.

Please check  http://ebootcamp.opensips.org/

-giovanni

On Thu, Jul 28, 2016 at 6:41 AM, Owais Ahmad <millennium.bug at gmail.com>
wrote:

> Hi Bogdan,
>
> Will this training session be webcast?
>
> Regards,
> Owais
>
> On Thu, Jul 28, 2016 at 2:51 AM, Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hello,
>>
>> The detailed description of the training content can be found here :
>>
>> http://www.opensips.org/pub/events/2016-08-12_OpenSIPS-ClueCon_Chicago/ClueConTraining.pdf
>>
>> See you in Chicago,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>> On 17.06.2016 17:35, Bogdan-Andrei Iancu wrote:
>>
>>> Hello,
>>>
>>> As part of ClueCon, we will run an one-day hands-on Training around
>>> OpenSIPS and FreeSWITCH integration. The training follows the steps of
>>> building a complete system featuring:
>>>     * OpenSIPS as a cluster front-end  - http://opensips.org
>>>     * FreeSWITCH PBX core system  - http://freeswitch.org/
>>>     * HOMER for SIP capturing  - http://sipcapture.org/
>>>     * CGRates as billing engine  - http://cgrates.org/
>>>     * SIP Fraud detection
>>>
>>> The training will be held on 12th of August 2016, as part of the ClueCon
>>> event.
>>>
>>> Together with the training, we will run a new session of Design Clinics (
>>> http://www.opensips.org/Community/Clinics)
>>>
>>> Details, registration (both training and clinics) and more are available
>>> here:
>>>     http://www.opensips.org/events/Training-2016ClueCon.html
>>>
>>>
>>> Best regards
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From goup2010 at gmail.com  Thu Jul 28 14:38:26 2016
From: goup2010 at gmail.com (Dragomir Haralambiev)
Date: Thu, 28 Jul 2016 15:38:26 +0300
Subject: [OpenSIPS-Users] Redis problem in Opensips 2.2.1
Message-ID: <CADzurO+OtH=v7Rqh7T-VneePo2ayAtBEXozMgzvNJ=y9O8Cquw@mail.gmail.com>

Hello ,

I try to setup Regis in opensips 2.2.1.

I put in config file follow lines:
loadmodule "cachedb_redis.so"
modparam("cachedb_redis", "cachedb_url","redis:group1://localhost:6379/");

When start Opensips I receive follow errors:

Jul 28 15:31:46 localhost opensips: DBG:core:load_module: loading module
/usr/local//lib64/opensips/modules/cachedb_redis.so
Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex:
cachedb_redis matches module cachedb_redis
Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex: found
<cachedb_url> in module cachedb_redis [/usr/local//lib64/opensips/modules/]
Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error in
config file /usr/local//etc/opensips/opensips.cfg, line 114, column 20-21:
syntax error
Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error in
config file /usr/local//etc/opensips/opensips.cfg, line 114, column 20-21:
Jul 28 15:31:46 localhost opensips: ERROR:core:main: bad config file (2
errors)


Where is problem?

Best regards,
Dragomir
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From bogdan at opensips.org  Wed Jul 27 23:56:29 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 28 Jul 2016 00:56:29 +0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
	<CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
Message-ID: <57992E0D.7020701@opensips.org>

Hi Cesar,

Be aware that all the MI commands (like dlg_list) can be triggered via 
different backends. Maybe doing it via mi_json will give you a better 
(and easy to parse) answer.
     http://www.opensips.org/html/docs/modules/2.2.x/mi_json.html

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 27.07.2016 22:57, Cesar Alberto Rodriguez Fierro wrote:
> There was a wrong configuration of the "dialog.so" module.
>
> I will try to parse the output in order to obtain the current calls.
>
>  Thanks for the help.
>
> On Wed, Jul 27, 2016 at 1:20 PM, Carlos Eduardo <kaduww at gmail.com 
> <mailto:kaduww at gmail.com>> wrote:
>
>     Cesar,
>
>     Are you using the dialog module in your script?
>
>     This MI command will only return a valid value if the dialog
>     module is loaded (loadmodule "dialog.so") and if the dialogs are
>     criated durign the script processing (create_dialog() command).
>
>     2016-07-27 16:12 GMT-03:00 Daniel Zanutti
>     <daniel.zanutti at gmail.com <mailto:daniel.zanutti at gmail.com>>:
>
>         On my sample, you should run:
>
>         opensipsctl fifo profile_get_size inbound
>
>
>         Using dlg_list you should get something like this:
>         # opensipsctl fifo dlg_list
>         dialog::  hash=84:1411689852
>                 state:: 4
>                 user_flags:: 0
>                 timestart:: 1469646635
>                 datestart:: 2016-07-27 16:10:35
>                 timeout:: 1469653835
>                 dateout:: 2016-07-27 18:10:35
>         ...
>         dialog::  hash=289:324429409
>                 state:: 2
>                 user_flags:: 0
>                 timestart:: 0
>                 timeout:: 0
>         ...
>         dialog::  hash=640:1695114669
>                 state:: 4
>         ...
>
>         Check if modules are successfully loaded.
>
>         Regards
>
>         On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez
>         Fierro <crf at transtelco.net <mailto:crf at transtelco.net>> wrote:
>
>
>             Thanks for your help.
>
>             I trying to use FIFO in order to  send requests to
>             OpenSIPS, I have read some documentation about the  Dialog
>             Module.  I guess using the "opensipsctl fifo dlg_list"
>             command can be useful to obtain the current calls, but I
>             am not sure why the command is not available in my
>             OpenSIPS version.   When I execute the
>             command ./opensipsctl fifo version, I am getting the
>             following information
>             Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>
>
>
>             On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho
>             <pimenta at inatel.br <mailto:pimenta at inatel.br>> wrote:
>
>                 Now, thinking more about it, I would suggest you to
>                 put a SQL query in your proprietary software to query
>                 the OpenSIPS database directly.
>
>                 The table dialog will be always updated about current
>                 calls.
>
>
>                 Regards.
>
>
>                 RODRIGO PIMENTA CARVALHO
>                 Inatel Competence Center
>                 Software
>                 Ph: +55 35 3471 9200 <tel:%2B55%2035%203471%209200>
>                 RAMAL 979
>
>
>                 ------------------------------------------------------------------------
>                 *De:* users-bounces at lists.opensips.org
>                 <mailto:users-bounces at lists.opensips.org>
>                 <users-bounces at lists.opensips.org
>                 <mailto:users-bounces at lists.opensips.org>> em nome de
>                 Rodrigo Pimenta Carvalho <pimenta at inatel.br
>                 <mailto:pimenta at inatel.br>>
>                 *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>                 *Para:* users at lists.opensips.org
>                 <mailto:users at lists.opensips.org>
>                 *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls
>                 from sip server.
>
>                 With FIFO you can send requests to OpenSIPS, for
>                 example from a proprietary software. So, if a request
>                 wants to execute a query with avpops, I guess FIFO
>                 will be useful.
>
>
>                 Regards.
>
>
>
>                 RODRIGO PIMENTA CARVALHO
>                 Inatel Competence Center
>                 Software
>                 Ph: +55 35 3471 9200 <tel:%2B55%2035%203471%209200>
>                 RAMAL 979
>
>
>                 ------------------------------------------------------------------------
>                 *De:* users-bounces at lists.opensips.org
>                 <mailto:users-bounces at lists.opensips.org>
>                 <users-bounces at lists.opensips.org
>                 <mailto:users-bounces at lists.opensips.org>> em nome de
>                 Cesar Alberto Rodriguez Fierro <crf at transtelco.net
>                 <mailto:crf at transtelco.net>>
>                 *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>                 *Para:* users at lists.opensips.org
>                 <mailto:users at lists.opensips.org>
>                 *Assunto:* [OpenSIPS-Users] Get concurrent calls from
>                 sip server.
>                 Hi !
>
>                 I am currently working in a project related with
>                 display in real time the active calls of our VoIP
>                 traffic, I would like to get the active sip-calls from
>                 a Kamailio Sip Server (running opensips), is there any
>                 way to obtain this information.
>
>                 Best Regards.
>
>
>
>
>                 Inline image 1|Cesar Rodriguez | VoiceOPS  | MX: +52
>                 656-257-4112 <tel:%2B52%20656-257-4112> |
>
>                 CONFIDENTIALITY NOTICE:  This communication is
>                 intended only for the use of the individual or entity
>                 to which it is addressed and may contain information
>                 that is privileged, confidential, and exempt from
>                 disclosure under applicable law.  If you are not the
>                 intended recipient of this information, you are
>                 notified that any use, dissemination, distribution, or
>                 copying of the communication is strictly prohibited.
>
>                 _______________________________________________
>                 Users mailing list
>                 Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>             _______________________________________________
>             Users mailing list
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>     -- 
>     *Carlos E. Wagner*
>     /Tecn?logo em Telecomunica??es, OCP, dCAA/
>     *
>     *
>     *Gnotel Telecom*
>     *E-mail:*_kaduww at gmail.com <mailto:kaduww at gmail.com>_
>     _carlos at gnotel.com.br <mailto:carlos at gnotel.com.br>_
>     *Fone:*+55 48 9981-0894 <tel:%2B55%2048%209981-0894>
>     *Skype:*carlos.e.wagner
>     www.gnotel.com.br <http://www.gnotel.com.br>
>     www.blogdovoip.com <http://www.blogdovoip.com>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From razvan at opensips.org  Thu Jul 28 14:53:32 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Thu, 28 Jul 2016 15:53:32 +0300
Subject: [OpenSIPS-Users] Redis problem in Opensips 2.2.1
In-Reply-To: <CADzurO+OtH=v7Rqh7T-VneePo2ayAtBEXozMgzvNJ=y9O8Cquw@mail.gmail.com>
References: <CADzurO+OtH=v7Rqh7T-VneePo2ayAtBEXozMgzvNJ=y9O8Cquw@mail.gmail.com>
Message-ID: <a4e71d38-b7da-9d19-827c-dfad53f30258@opensips.org>

Hi, Dragomir!

You have a ";" at the end of the modparam line. Please remove it and try 
again.

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/28/2016 03:38 PM, Dragomir Haralambiev wrote:
> Hello ,
>
> I try to setup Regis in opensips 2.2.1.
>
> I put in config file follow lines:
> loadmodule "cachedb_redis.so"
> modparam("cachedb_redis", "cachedb_url","redis:group1://localhost:6379/");
>
> When start Opensips I receive follow errors:
>
> Jul 28 15:31:46 localhost opensips: DBG:core:load_module: loading 
> module /usr/local//lib64/opensips/modules/cachedb_redis.so
> Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex: 
> cachedb_redis matches module cachedb_redis
> Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex: 
> found <cachedb_url> in module cachedb_redis 
> [/usr/local//lib64/opensips/modules/]
> Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error 
> in config file /usr/local//etc/opensips/opensips.cfg, line 114, column 
> 20-21: syntax error
> Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error 
> in config file /usr/local//etc/opensips/opensips.cfg, line 114, column 
> 20-21:
> Jul 28 15:31:46 localhost opensips: ERROR:core:main: bad config file 
> (2 errors)
>
>
> Where is problem?
>
> Best regards,
> Dragomir
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From goup2010 at gmail.com  Thu Jul 28 15:05:29 2016
From: goup2010 at gmail.com (Dragomir Haralambiev)
Date: Thu, 28 Jul 2016 16:05:29 +0300
Subject: [OpenSIPS-Users] Redis problem in Opensips 2.2.1
In-Reply-To: <a4e71d38-b7da-9d19-827c-dfad53f30258@opensips.org>
References: <CADzurO+OtH=v7Rqh7T-VneePo2ayAtBEXozMgzvNJ=y9O8Cquw@mail.gmail.com>
	<a4e71d38-b7da-9d19-827c-dfad53f30258@opensips.org>
Message-ID: <CADzurOL0ZyZSFO3vnhb92Mekpo512gq0+VStmbzSvuewSZ15Gw@mail.gmail.com>

YES now all is OK.
Thanks!

2016-07-28 15:53 GMT+03:00 R?zvan Crainea <razvan at opensips.org>:

> Hi, Dragomir!
>
> You have a ";" at the end of the modparam line. Please remove it and try
> again.
>
> Best regards,
>
> R?zvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 07/28/2016 03:38 PM, Dragomir Haralambiev wrote:
>
> Hello ,
>
> I try to setup Regis in opensips 2.2.1.
>
> I put in config file follow lines:
> loadmodule "cachedb_redis.so"
> modparam("cachedb_redis", "cachedb_url","redis:group1://localhost:6379/");
>
> When start Opensips I receive follow errors:
>
> Jul 28 15:31:46 localhost opensips: DBG:core:load_module: loading module
> /usr/local//lib64/opensips/modules/cachedb_redis.so
> Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex:
> cachedb_redis matches module cachedb_redis
> Jul 28 15:31:46 localhost opensips: DBG:core:set_mod_param_regex: found
> <cachedb_url> in module cachedb_redis [/usr/local//lib64/opensips/modules/]
> Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error in
> config file /usr/local//etc/opensips/opensips.cfg, line 114, column 20-21:
> syntax error
> Jul 28 15:31:46 localhost opensips: CRITICAL:core:yyerror: parse error in
> config file /usr/local//etc/opensips/opensips.cfg, line 114, column 20-21:
> Jul 28 15:31:46 localhost opensips: ERROR:core:main: bad config file (2
> errors)
>
>
> Where is problem?
>
> Best regards,
> Dragomir
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
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From pwakano at gmail.com  Thu Jul 28 15:38:08 2016
From: pwakano at gmail.com (Patrick Wakano)
Date: Thu, 28 Jul 2016 10:38:08 -0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
	<CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
Message-ID: <CAPu3kNVS6jV_weKfW_N1pqUszSGxAC_wvh07YMERCCC9=Y-RDA@mail.gmail.com>


From Ben.Newlin at inin.com  Thu Jul 28 16:13:40 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Thu, 28 Jul 2016 14:13:40 +0000
Subject: [OpenSIPS-Users] How to tell if uac_auth() is successful
In-Reply-To: <579867EC.7030609@opensips.org>
References: <9A19D7A7-A0A9-4D15-8586-3DC345185167@inin.com>
	<579867EC.7030609@opensips.org>
Message-ID: <D8A8AF1D-BFDB-4F08-AFAF-8A4CF363B023@inin.com>

Bogdan,

You are right. I must have made a mistake in my previous test. Thanks!

The module documentation does not mention that the function returns a result.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:51 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] How to tell if uac_auth() is successful

Hi Ben

The uac_auth() should return true if the authentication was successfully performed. Are you sure you correctly perform the test on the return code? Do you see any error log from the function ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 25.07.2016 23:00, Newlin, Ben wrote:
I am using the UAC and UAC_AUTH modules to perform trunk authentication. The uac_auth() function will only use credentials when the realm matches one in the 401/407 challenge. Is there any way to tell whether there was a successful authentication match?

The function doesn?t provide any return code (returns false even if match was found). I have tried checking for values in Authorization header ($ar, $auth.resp) but they are not populated.

If I can?t tell whether appropriate credentials were found, then I must always send another INVITE which might not have any Authorization and will just be rejected again.

Any help would be appreciated.

Thanks,
Ben Newlin




_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Jul 29 12:39:26 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 29 Jul 2016 13:39:26 +0300
Subject: [OpenSIPS-Users] How to tell if uac_auth() is successful
In-Reply-To: <D8A8AF1D-BFDB-4F08-AFAF-8A4CF363B023@inin.com>
References: <9A19D7A7-A0A9-4D15-8586-3DC345185167@inin.com>
	<579867EC.7030609@opensips.org>
	<D8A8AF1D-BFDB-4F08-AFAF-8A4CF363B023@inin.com>
Message-ID: <579B325E.1020802@opensips.org>

Hi Ben,

All functions have a return code. If there is nothing particulary 
mentioned, the stanard interpretation of the return codesis:
     false ( <0) - function failed
     true (>0) - function succeeded

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 28.07.2016 17:13, Newlin, Ben wrote:
>
> Bogdan,
>
> You are right. I must have made a mistake in my previous test. Thanks!
>
> The module documentation does not mention that the function returns a 
> result.
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Wednesday, July 27, 2016 at 3:51 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>, 
> "Newlin, Ben" <Ben.Newlin at inin.com>
> *Subject: *Re: [OpenSIPS-Users] How to tell if uac_auth() is successful
>
> Hi Ben
>
> The uac_auth() should return true if the authentication was 
> successfully performed. Are you sure you correctly perform the test on 
> the return code? Do you see any error log from the function ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 25.07.2016 23:00, Newlin, Ben wrote:
>
>     I am using the UAC and UAC_AUTH modules to perform trunk
>     authentication. The uac_auth() function will only use credentials
>     when the realm matches one in the 401/407 challenge. Is there any
>     way to tell whether there was a successful authentication match?
>
>     The function doesn?t provide any return code (returns false even
>     if match was found). I have tried checking for values in
>     Authorization header ($ar, $auth.resp) but they are not populated.
>
>     If I can?t tell whether appropriate credentials were found, then I
>     must always send another INVITE which might not have any
>     Authorization and will just be rejected again.
>
>     Any help would be appreciated.
>
>     Thanks,
>
>     Ben Newlin
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From razvan at opensips.org  Fri Jul 29 13:07:53 2016
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 29 Jul 2016 14:07:53 +0300
Subject: [OpenSIPS-Users] name of event in the event message's body
In-Reply-To: <1003271469625183@web21h.yandex.ru>
References: <1003271469625183@web21h.yandex.ru>
Message-ID: <89fa155e-5952-e839-2c48-344ffdd40bcd@opensips.org>

Hi!

Unfortunately there currently is no mechanism for not sending the 
event's name. All you can do is discard the data from the application 
that receives the event.

If you find useful such a feature, please open a feature request on our 
tracker[1], and we will try to find a solution.

[1] https://github.com/OpenSIPS/opensips/issues

Best regards,

R?zvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com

On 07/27/2016 04:13 PM, ?????? ????? wrote:
>
> Hi all!
>
> I trying to send some data by event_interface via rabbitmq. For do 
> that, i use subscribe_event("EVENT_NAME", "rabbitmq:someaddress"), and 
> then raise_event("EVENT_NAME", $var(eventdata)).All works fine, but 
> the body of message, sent by raise_event, looks like "EVENT_NAME 
> eventdata", but i need to send only eventdata, without EVENT_NAME. Is 
> there some way to not send name of raised event in the body of message?
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Fri Jul 29 14:40:33 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 29 Jul 2016 15:40:33 +0300
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
Message-ID: <579B4EC1.4010809@opensips.org>

Hi Ben,

Sorry to disagree, but IMHO they do exclude one eachother.

Adding RR to TH should not be seen as a way of fixing some broken TH 
scenarios (with advertise).
So,let me try to understand what is not working for you.You do TH and 
advertise. In this case, normally, in the Contact headers generated by 
OpenSIPS (as a result of TH), it should be the TH interface, right ? 
What exactly seems to be the problem ? do you have a trace to show the 
issues ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 27.07.2016 16:05, Newlin, Ben wrote:
>
> I understand that normally you would not need RR with TH, but the two 
> concepts are not mutually exclusive in SIP. As I said, I have a need 
> to Record-Route the call on my server as I am advertising a different 
> address than I am listening on. This means that TH will populate the 
> Contact header with the advertised address and if I cannot 
> Record-Route with the actual address then I will not receive 
> sequential requests.
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Wednesday, July 27, 2016 at 3:59 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>, 
> "Newlin, Ben" <Ben.Newlin at inin.com>
> *Subject: *Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
>
> Hi Ben,
>
> As I mentioned in different thread, TH is not compatible with the RR 
> mechanism. If you do TH, your OpenSIPS will act as and end point (from 
> SIP perspective), so there will be no Route/RR headers at all. So no 
> need to do loose_route or so. You just do TH matching for the 
> sequential requests and nothing more.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 22.07.2016 16:48, Newlin, Ben wrote:
>
>     Hi,
>
>     I am using the Dialog module with topology_hiding() in my server
>     and I have a need to Record-Route the call on my server as I am
>     advertising a different address than I am listening on. I have
>     found what I believe is an inconsistency in the handling of
>     Record-Route within the Dialog topology_hiding functionality. The
>     topology_hiding isn?t a true B2BUA, but it does set up different
>     parameters for the incoming UAC and outgoing UAS sides of the call
>     for the Via headers, Record-Route and Route headers, and the
>     Contact header(s).
>
>     The problem is that the record_route() and loose_route() functions
>     operate on different sides of the call. The record_route()
>     function will only add a Record-Route header to the outgoing UAS
>     side of the call. And since the record_route() function cannot be
>     called from onreply_route, but is no way to add a Record-Route
>     header to the UAC side of the call.
>
>     On the other hand, the loose_route() function only operates on the
>     incoming UAC side of the call and there is no way to perform
>     loose_route() on the UAS side of the call.
>
>     So there is a situation where Record-Route headers can only be
>     added on the outgoing UAS side, but the associated Route headers
>     can only be removed on the incoming UAC side (where they won?t
>     exist since they can?t be added) and any added headers on the UAS
>     side cannot be processed properly due to the lack of loose_route.
>
>     I can provide further information if this is unclear. It should be
>     easily reproducible by attempting to use record_route in a
>     topology_hiding scenario. The route is added to the outbound leg,
>     but is not removed by loose_route so the message is looped back
>     every time.
>
>     *Ben Newlin***| Sr Voice Network Engineer, PureCloud
>
>     phone & fax +1.317.957.1009 | ben.newlin at inin.com
>     <mailto:ben.newlin at inin.com>
>
>     mage removed by sender.
>
>     www.inin.com <http://www.inin.com>
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Fri Jul 29 14:45:42 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 29 Jul 2016 15:45:42 +0300
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <4bfa207a853f49eba59637064b77e650@COPDCEX28.cable.comcast.com>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
	<6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
	<57986130.2030505@opensips.org>
	<4bfa207a853f49eba59637064b77e650@COPDCEX28.cable.comcast.com>
Message-ID: <579B4FF6.3040603@opensips.org>

Hi Agalya,

Inthe onreply_route(), the logic seems ok:
     * if flag set -> drop and log
     * for all 180, set the flag.
This should allow only the first 180 reply to pass.

But I do not understand what is the purpose of the relay2 route? the 
onreply route should go everything you need (in terms of 180 logic).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 27.07.2016 16:27, Ramachandran, Agalya (Contractor) wrote:
>
> Hi Bogdan,
>
> Yes you are correct. Am trying to implement that logic. (keep and log 
> the first 180, drop the others)
>
> Am giving you snippet of onreply_route and relay2 attached here with 
> this mail.
>
> My intention is just to log the response sent by OpenSIPS. For other 
> responses logic seems good and I could see that response sent out are 
> logged.
>
> Only for 180 resposne, am not meeting my condition.
>
> Correct me if am wrong.
>
> Regards,
>
> Agalya
>
> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
> *Sent:* Wednesday, July 27, 2016 3:22 AM
> *To:* Ramachandran, Agalya (Contractor) 
> <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list 
> <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] Provisional response handling in case 
> of forking
>
> Hi Agalya,
>
> First, take care that you can set and trigger only one reply route per 
> transaction !
>
> Now, looking at your logic in the script, it does not reflect your 
> statement : keep and log the first 180, drop the others <- is this 
> correct ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
>
>     Hi Bogdan,
>
>     I have a question related to the below topic.
>
>     When am receiving more than  180 response from different
>     destinations , am sending 1^st 180 to the caller and dropping
>     consecutive  180 ?s.
>
>     And this logic is working fine.
>
>     Now I have a scenario, where I need to log what are the responses
>     sent out of Opensips.
>
>     My logic for this is, am checking the flag value. If flag is set,
>     log the 1^st 180 response, reset the flag and drop the remaining
>     180 response.
>
>     I could see FLAG_180 is set, and when I try to reset flag *is not
>     getting reset*. Is it not possible to reset the flag values when
>     relay is done?
>
>     I have added a route(relay2) in onreply_route  Below is the
>     snippet of it.
>
>     Please guide me if am doing something wrong.
>
>     route[relay2] {
>
>             if ($rs==100)
>
>     drop();
>
>             if ($rs==180) {
>
>               if (isflagset(FLAG_180)){
>
>     xlog("INFO:opensips: Flag is set \n" <INFO:opensips:Flagisset%5Cn>);
>
>     resetflag(FLAG_180);
>
>               }
>
>               else{
>
>          drop();
>
>              }
>
>     #setflag(FLAG_180);
>
>             }
>
>             xlog("INFO:opensips: Sending $rs resposne out \n"
>     <INFO:opensips:Sending$rsresposneout%5Cn>);
>
>     }
>
>     Regards,
>     Agalya
>
>     *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
>     *Sent:* Monday, June 13, 2016 6:31 AM
>     *To:* OpenSIPS users mailling list <users at lists.opensips.org>
>     <mailto:users at lists.opensips.org>; Ramachandran, Agalya
>     (Contractor) <Agalya_Ramachandran at comcast.com>
>     <mailto:Agalya_Ramachandran at comcast.com>
>     *Subject:* Re: [OpenSIPS-Users] Provisional response handling in
>     case of forking
>
>     Hi Agalya,
>
>     Use the onreply route (be sure to onreply_avp_mode to be set to 1
>     - see
>     http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290)
>     in combination with flags, to record when the an 180 reply was set.
>
>     Like:
>
>     if ($rs==180) {
>         if (isflagset(FLAG_180))
>             drop();
>     setflag(FALG_180);
>     }
>
>     The onreply_avp_mode 1 will ensure that the onreply route will not
>     overlap for 2 replies .
>
>     Regards,
>
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>
>     http://www.opensips-solutions.com
>
>     On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
>
>         Hi team,
>
>         We are using opensips for our project and we are currently
>         using opensips as proxy.
>
>         Am forking the incoming sip call, to two destinations. It
>         Rings in both Dest A and Dest B, as a result I get two 180
>         Ringing response from A and B.
>
>         I want to filter only the first incoming 180 Ringing response
>         and send to the actual caller. Is there a way to do this in
>         opensips config file?
>
>         I have seen drop() function which drops the complete
>         provisional response. But in my case I have to forward one 180
>         Ringing to the caller.
>
>         Can it be achieved by the changes in config file? Please guide me.
>
>         Regards,
>         Agalya
>
>
>
>
>
>         _______________________________________________
>
>         Users mailing list
>
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From bogdan at opensips.org  Fri Jul 29 14:50:47 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 29 Jul 2016 15:50:47 +0300
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
Message-ID: <579B5127.7080805@opensips.org>

Ben,

What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 27.07.2016 19:02, Newlin, Ben wrote:
>
> I have identified that these crashes are occurring when the far end 
> system is not returning the Record-Route headers in the 200 OK 
> response. The headers are present in the 180 response, but not the 200 
> OK. I have reproduced the scenario using SIPp and captured a SIP 
> trace: http://pastebin.com/ckKk3EhY <http://pastebin.com/ckKk3EhY>
>
> The crash occurs on receipt of the ACK request and attempt to match 
> the dialog.
>
> I also captured a BT for this scenario as well, in case anything 
> specific in the trace made the issue easier to find: 
> http://pastebin.com/cM3FhPiw
>
> I am working with the other system to try to fix their behavior.
>
> Ideally the Record-Route headers from previous replies could be used 
> in this case to allow the call to succeed, but I don?t know if that is 
> possible.
>
> Thanks,
>
> Ben Newlin
>
> *From: *"Newlin, Ben" <Ben.Newlin at inin.com>
> *Date: *Wednesday, July 27, 2016 at 9:44 AM
> *To: *Bogdan-Andrei Iancu <bogdan at opensips.org>, OpenSIPS users 
> mailling list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
> Bogdan,
>
> This is a different scenario than the other you responded to. As I 
> said, we have two types of servers that work together. One is a 
> load-balancer and runs as a proxy. It uses double Record-Route because 
> it sends messages between public and private networks. Then we have 
> our other servers using TH which receive those requests. We are not 
> using TH and RR on the same server (although I would like to).
>
> If validate_dialog() and fix_route_dialog() (and possibly 
> loose_route()) should not be called when using TH, I believe the 
> documentation should reference that. It states that match_dialog() 
> must be used with TH, but does not indicate that the other functions 
> should not be used or that the functionality won?t work. There is also 
> no documentation of the incompatibility between RR and TH.
>
> Either way, I ran a test where I removed all calls to loose_route(), 
> validate_dialog(), and fix_route_dialog() from my script. The crash 
> still occurred and the BT still pointed to fix_route_dialog() 
> function. So it must be getting called from within Dialog module 
> somewhere. That BT is here: http://pastebin.com/wu2X2Hxh
>
> I collected this BT with loose_route() being called from my script, 
> but not validate_dialog() or fix_route_dialog(): 
> http://pastebin.com/6V7yPaHF
>
> This BT was collected with all three functions being called from my 
> script: http://pastebin.com/fZYYdndn
>
> Ben Newlin
>
> *From: *Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Date: *Wednesday, July 27, 2016 at 3:57 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>, 
> "Newlin, Ben" <Ben.Newlin at inin.com>
> *Subject: *Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
>
> Hi Ben,
>
> First, if you use TH, makes no sense to do Record-Routing - there are 
> 2 SIP concepts that overlaps. You either act as an end-point (by doing 
> TH), either as a proxy (doing RR).
>
> If doing TH, makes no sense to use validate + fix as these functions 
> check and repair the routing information in the request (like Route 
> and Contact headers). if you do TH, this routing info is actually 
> hidden and added by OpenSIPS, so there is nothing to fix and repair.
>
> Nevertheless, this should not crash or corrupt OpenSIPS. HAve you 
> managed to get a corefile ?
>
> Also if you suspect memory corruption, you can compile-in the memory 
> debugger - see 
> http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .
>
> Regards,
>
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> On 26.07.2016 23:20, Newlin, Ben wrote:
>
>     I have had 3 OpenSIPS server crashes in the last week. All were
>     due to segmentation faults. I was not able to capture core dumps;
>     I am configuring that now to catch the next crash.
>
>     My logs leading up to the crash are full of errors from
>     fix_route_dialog() complaining about invalid URIs for sequential
>     requests:
>
>     Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to
>     parse SIP uri
>
>     Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0
>     parsed: <ip:1> (4) /
>     <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)
>
>     Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to
>     parse SIP uri
>
>     Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0
>     parsed: <b0i2> (4) /
>     <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)
>
>     Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to
>     parse SIP uri
>
>     Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0
>     parsed: <ervi> (4) /
>     <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)
>
>     Many times the ?URI? displayed in the error message is actually
>     internal OpenSIPS variables, as in the last error above. When they
>     are from the SIP message, I have verified that the messages
>     themselves are correctly formatted. This leads me to believe there
>     is memory corruption occurring.
>
>     This all started when I updated my load-balancer servers to use
>     Record-Routing, specifically the ?double_rr? mechanism for when
>     multiple interfaces exist. The Record-Routing is occurring on
>     different servers which have not crashed. Only the servers
>     receiving the Record-Routed messages are experiencing the errors.
>
>     Here is a piece of the code processing sequential requests. I am
>     using the topology_hiding() functionality of the Dialog module.
>     Are validate_dialog() and fix_route_dialog() still valid in a
>     topology_hiding scenario?
>
>     if (t_check_trans())
>
>     setflag(SEQ_REQUEST);
>
>     if (has_totag())
>
>     {
>
>     loose_route();
>
>     if (match_dialog())
>
>     {
>
>     if (!validate_dialog())
>
>     fix_route_dialog();
>
>     if (is_method("BYE"))
>
>     setflag(ACC_FLAG);
>
>     setflag(SEQ_REQUEST);
>
>     }
>
>     else if (!isflagset(SEQ_REQUEST))
>
>     {
>
>     if (!is_method("ACK")) {
>
>     route(rlog, LV_ERROR, "check_sequential", "Sequential request not
>     matched");
>
>       route(reply_error, "481", "Call Does Not Exist");
>
>     }
>
>     return(EXIT);
>
>     }
>
>     }
>
>     I will attempt to get core dumps of future crashes.
>
>     Thanks,
>
>     Ben Newlin
>
>
>
>
>
>     _______________________________________________
>
>     Users mailing list
>
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

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From Ben.Newlin at inin.com  Fri Jul 29 14:54:13 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Fri, 29 Jul 2016 12:54:13 +0000
Subject: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes
In-Reply-To: <579B5127.7080805@opensips.org>
References: <885977D6-9A91-45A9-8BA6-EC6369CD45B8@inin.com>
	<5798697D.8070407@opensips.org>
	<7B64399E-7EAF-40EC-B580-BF6ABBE32BA7@inin.com>
	<204A9049-5F70-4462-B6E6-1E00F59D8088@inin.com>
	<579B5127.7080805@opensips.org>
Message-ID: <273E1DB2-DD7E-412E-B762-2E1826B2AC7D@inin.com>

This is 1.11.6, running on CentOS 7.


Ben Newlin

From: <users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org>
Reply-To: OpenSIPS users mailling list <users at lists.opensips.org>
Date: Friday, July 29, 2016 at 8:50 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Ben,

What OpenSIPS version is this (the crashing one) ? 1.11 or 2.1 ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 19:02, Newlin, Ben wrote:
I have identified that these crashes are occurring when the far end system is not returning the Record-Route headers in the 200 OK response. The headers are present in the 180 response, but not the 200 OK. I have reproduced the scenario using SIPp and captured a SIP trace: http://pastebin.com/ckKk3EhY

The crash occurs on receipt of the ACK request and attempt to match the dialog.

I also captured a BT for this scenario as well, in case anything specific in the trace made the issue easier to find: http://pastebin.com/cM3FhPiw

I am working with the other system to try to fix their behavior.

Ideally the Record-Route headers from previous replies could be used in this case to allow the call to succeed, but I don?t know if that is possible.

Thanks,

Ben Newlin

From: "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Date: Wednesday, July 27, 2016 at 9:44 AM
To: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>, OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Bogdan,

This is a different scenario than the other you responded to. As I said, we have two types of servers that work together. One is a load-balancer and runs as a proxy. It uses double Record-Route because it sends messages between public and private networks. Then we have our other servers using TH which receive those requests. We are not using TH and RR on the same server (although I would like to).

If validate_dialog() and fix_route_dialog() (and possibly loose_route()) should not be called when using TH, I believe the documentation should reference that. It states that match_dialog() must be used with TH, but does not indicate that the other functions should not be used or that the functionality won?t work. There is also no documentation of the incompatibility between RR and TH.

Either way, I ran a test where I removed all calls to loose_route(), validate_dialog(), and fix_route_dialog() from my script. The crash still occurred and the BT still pointed to fix_route_dialog() function. So it must be getting called from within Dialog module somewhere. That BT is here: http://pastebin.com/wu2X2Hxh

I collected this BT with loose_route() being called from my script, but not validate_dialog() or fix_route_dialog(): http://pastebin.com/6V7yPaHF

This BT was collected with all three functions being called from my script: http://pastebin.com/fZYYdndn


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:57 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] OpenSIPS fix_route_dialog crashes

Hi Ben,

First, if you use TH, makes no sense to do Record-Routing - there are 2 SIP concepts that overlaps. You either act as an end-point (by doing TH), either as a proxy (doing RR).

If doing TH, makes no sense to use validate + fix as these functions check and repair the routing information in the request (like Route and Contact headers). if you do TH, this routing info is actually hidden and added by OpenSIPS, so there is nothing to fix and repair.

Nevertheless, this should not crash or corrupt OpenSIPS. HAve you managed to get a corefile ?

Also if you suspect memory corruption, you can compile-in the memory debugger - see http://www.opensips.org/Documentation/TroubleShooting-OutOfMem .

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 23:20, Newlin, Ben wrote:
I have had 3 OpenSIPS server crashes in the last week. All were due to segmentation faults. I was not able to capture core dumps; I am configuring that now to catch the next crash.

My logs leading up to the crash are full of errors from fix_route_dialog() complaining about invalid URIs for sequential requests:

Jul 26 19:34:02 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:34:02 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ip:1> (4) / <ip:10.18.8.18:5060;ftag=gK0448f137;lr;r2=on>> (44)

Jul 26 19:11:19 [218] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 19:11:19 [218] ERROR:core:parse_uri: bad uri, state 0 parsed: <b0i2> (4) / <b0i2yjor;transport=udp<sip:10.18.8.17:5060;ftag=7207ce89;lr;r2=on> (65)

Jul 26 17:43:19 [220] ERROR:dialog:fix_route_dialog: Failed to parse SIP uri
Jul 26 17:43:19 [220] ERROR:core:parse_uri: bad uri, state 0 parsed: <ervi> (4) / <ervice_id6fdbc70f-2438-4726-807c-0d081df4d87> (44)

Many times the ?URI? displayed in the error message is actually internal OpenSIPS variables, as in the last error above. When they are from the SIP message, I have verified that the messages themselves are correctly formatted. This leads me to believe there is memory corruption occurring.

This all started when I updated my load-balancer servers to use Record-Routing, specifically the ?double_rr? mechanism for when multiple interfaces exist. The Record-Routing is occurring on different servers which have not crashed. Only the servers receiving the Record-Routed messages are experiencing the errors.

Here is a piece of the code processing sequential requests. I am using the topology_hiding() functionality of the Dialog module. Are validate_dialog() and fix_route_dialog() still valid in a topology_hiding scenario?

if (t_check_trans())
    setflag(SEQ_REQUEST);

  if (has_totag())
  {
    loose_route();

    if (match_dialog())
    {
      if (!validate_dialog())
        fix_route_dialog();

      if (is_method("BYE"))
        setflag(ACC_FLAG);

      setflag(SEQ_REQUEST);
    }
    else if (!isflagset(SEQ_REQUEST))
    {
      if (!is_method("ACK")) {
        route(rlog, LV_ERROR, "check_sequential", "Sequential request not matched");
        route(reply_error, "481", "Call Does Not Exist");
      }

      return(EXIT);
    }
  }

I will attempt to get core dumps of future crashes.

Thanks,
Ben Newlin






_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From shahid.ge4 at gmail.com  Fri Jul 29 14:58:16 2016
From: shahid.ge4 at gmail.com (Shahid Mehmood)
Date: Fri, 29 Jul 2016 17:58:16 +0500
Subject: [OpenSIPS-Users] Get Error while send multiple value from 302 reply
Message-ID: <CANDZiCRhw5Rte774_zFD_MtVDex1c9211YUVgdqFgyyNWN3p-Q@mail.gmail.com>

I am getting  errors while returning multipule value from response of 302
Jul 28 21:58:01 Kamailio-2 /usr/sbin/kamailio[3939]: ERROR: pv
[pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
Jul 28 21:58:01 Kamailio-2 /usr/sbin/kamailio[3939]: ERROR: <core>
[sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
Jul 28 21:58:03 Kamailio-2 /usr/sbin/kamailio[3940]: ERROR: pv
[pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
Jul 28 21:58:03 Kamailio-2 /usr/sbin/kamailio[3940]: ERROR: <core>
[sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
Jul 28 21:58:07 Kamailio-2 /usr/sbin/kamailio[3956]: ERROR: pv
[pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
Jul 28 21:58:07 Kamailio-2 /usr/sbin/kamailio[3956]: ERROR: <core>
[sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
Jul 28 21:58:15 Kamailio-2 /usr/sbin/kamailio[3955]: ERROR: pv
[pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
Jul 28 21:58:15 Kamailio-2 /usr/sbin/kamailio[3955]: ERROR: <core>
[sr_module.c:1688]: get_str_fparam(): Could not convert PV to str
Jul 28 21:58:31 Kamailio-2 /usr/sbin/kamailio[3954]: ERROR: pv
[pv_core.c:241]: pv_get_ruri(): failed to parse the R-URI
Jul 28 21:58:31 Kamailio-2 /usr/sbin/kamailio[3954]: ERROR: <core>
[sr_module.c:1688]: get_str_fparam(): Could not convert PV to str



*and code is *


$var(x)= "sip:112233#923224210048 at 192.168.12.13";
$var(y)=$var(x) + ", "+ "sip:1122#923224210048 at 192.168.12.13";
$ruri=$var(y);
 sl_send_reply("302", "$ruri");



how can i remove this error?

-- 
Shahid Mehmood
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From Agalya_Ramachandran at comcast.com  Fri Jul 29 15:27:12 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Fri, 29 Jul 2016 13:27:12 +0000
Subject: [OpenSIPS-Users] Provisional response handling in case of
	forking
In-Reply-To: <579B4FF6.3040603@opensips.org>
References: <0686390e80e344cf9c2cff77c7b1dfd1@COPDCEX28.cable.comcast.com>
	<575E8B4E.2000708@opensips.org>
	<6e6aa2109da149e287131ad51155e89c@COPDCEX28.cable.comcast.com>
	<57986130.2030505@opensips.org>
	<4bfa207a853f49eba59637064b77e650@COPDCEX28.cable.comcast.com>
	<579B4FF6.3040603@opensips.org>
Message-ID: <ae62761d873f46c9b1229971c3a04a90@COPDCEX28.cable.comcast.com>

Hi Bogdan,

In onreply_route(), it logs all the incoming response to OpenSIPS.
My intention is to log and print the outgoing response from OpenSIPS.

A -------->OpenSIPS------->B

Here when User A calling User B, via OpenSIPS proxy, I am logging response am getting from User B.
I also want to log the response sent from OpenSIPS to User A.

As per my understanding , to log the response I have to do in source code from where the response is sent.  Due to some formatting we need for the project it is easy if I handle it in config file. So am calling relay2(route) and printing the logs.

Now please guide me what am doing is right or not, if not let me know what could be the best way to handle this.?

P.S : I want to log something like this.
xlog("INFO: event=Reply_from_client; reason=$rr; status=$rs [from=$fu;fromtag=$ft;to=$tu;totag=$tt;callId=$ci]\n");

Regards,
Agalya

From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Friday, July 29, 2016 8:46 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

In the onreply_route(), the logic seems ok:
    * if flag set -> drop and log
    * for all 180, set the flag.
This should allow only the first 180 reply to pass.

But I do not understand what is the purpose of the relay2 route? the onreply route should go everything you need (in terms of 180 logic).

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 16:27, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

Yes you are correct. Am trying to implement that logic. (keep and log the first 180, drop the others)
Am giving you snippet of onreply_route and relay2 attached here with this mail.

My intention is just to log the response sent by OpenSIPS. For other responses logic seems good and I could see that response sent out are logged.
Only for 180 resposne, am not meeting my condition.
Correct me if am wrong.

Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Wednesday, July 27, 2016 3:22 AM
To: Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>; OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

First, take care that you can set and trigger only one reply route per transaction !

Now, looking at your logic in the script, it does not reflect your statement : keep and log the first 180, drop the others <- is this correct ?

Best regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 26.07.2016 21:26, Ramachandran, Agalya (Contractor) wrote:
Hi Bogdan,

I have a question related to the below topic.
When am receiving more than  180 response from different destinations , am sending 1st 180 to the caller and dropping consecutive  180 's.
And this logic is working fine.

Now I have a scenario, where I need to log what are the responses sent out of Opensips.
My logic for this is, am checking the flag value. If flag is set, log the 1st 180 response, reset the flag and drop the remaining 180 response.

I could see FLAG_180 is set, and when I try to reset flag is not getting reset. Is it not possible to reset the flag values when relay is done?
I have added a route(relay2) in onreply_route  Below is the snippet of it.
Please guide me if am doing something wrong.

route[relay2] {
        if ($rs==100)
           drop();
        if ($rs==180) {
          if (isflagset(FLAG_180)){
                xlog("INFO:opensips: Flag is set \n"<INFO:opensips:Flagisset%5Cn>);
                resetflag(FLAG_180);
          }
          else{
              drop();
         }
          #setflag(FLAG_180);
        }
        xlog("INFO:opensips: Sending $rs resposne out \n"<INFO:opensips:Sending$rsresposneout\n>);
}


Regards,
Agalya
From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org]
Sent: Monday, June 13, 2016 6:31 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Ramachandran, Agalya (Contractor) <Agalya_Ramachandran at comcast.com><mailto:Agalya_Ramachandran at comcast.com>
Subject: Re: [OpenSIPS-Users] Provisional response handling in case of forking

Hi Agalya,

Use the onreply route (be sure to onreply_avp_mode to be set to 1 - see http://www.opensips.org/html/docs/modules/2.2.x/tm.html#id294290) in combination with flags, to record when the an 180 reply was set.

Like:

if ($rs==180) {
    if (isflagset(FLAG_180))
        drop();
    setflag(FALG_180);
}

The onreply_avp_mode 1 will ensure that the onreply route will not overlap for 2 replies .

Regards,




Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 10.06.2016 23:07, Ramachandran, Agalya (Contractor) wrote:
Hi team,

We are using opensips for our project and we are currently using opensips as proxy.
Am forking the incoming sip call, to two destinations. It Rings in both Dest A and Dest B, as a result I get two 180 Ringing response from A and B.
I want to filter only the first incoming 180 Ringing response and send to the actual caller. Is there a way to do this in opensips config file?

I have seen drop() function which drops the complete provisional response. But in my case I have to forward one 180 Ringing to the caller.
Can it be achieved by the changes in config file? Please guide me.

Regards,
Agalya







_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users



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From Ben.Newlin at inin.com  Fri Jul 29 17:05:02 2016
From: Ben.Newlin at inin.com (Newlin, Ben)
Date: Fri, 29 Jul 2016 15:05:02 +0000
Subject: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()
In-Reply-To: <579B4EC1.4010809@opensips.org>
References: <99A0B3CF-E6DF-46B3-BB9F-D953F0D043B3@inin.com>
	<579869F8.9060208@opensips.org>
	<105C468B-B9B5-4A03-899E-254F0FE1A2C9@inin.com>
	<579B4EC1.4010809@opensips.org>
Message-ID: <418CE841-BB2E-46D9-B8C3-2CD7BCDD5841@inin.com>

Here is the scenario:

My servers are only listening on a private IP address. There is a public address on our SBC. I have a carrier that requires that the Contact IP address matches the public address we provided to them. So when I do TH on my server I have to also do set_advertised_address to advertise the public address in the Contact header. Sequential requests use the Contact as the Request URI and the SBC is doing RR so all requests will come back through it. When the SBC receives a sequential request it strips its Route headers and forwards to the Request URI (previous Contact URI). But that URI now points back to the SBC, so it cannot deliver the request. If my server could Record-Route the initial request with its private address, the SBC would be able to route the request back to it.

Here is a trace of my scenario: http://pastebin.com/x927mFtq. I created it with SIPp so some endpoints are on the same IPs but with different ports. The public IP is 192.168.99.100, with port 7060 representing the PSTN carrier and port 5060 the SBC. The private IP is 10.0.2.15, with port 5060 being the SBC again and port 6060 being my server. You can see that the ACK cannot be delivered correctly to my server with this configuration.

The problem here stems from the fact that Topology Hiding should really be done at the edge of the network in order to be most effective. But my OpenSIPS server doesn?t sit on the Edge of the network, it is behind an SBC. So TH is complicated by the fact that my OpenSIPS has no public IP of its own and must advertise the public IP of the SBC instead, but future requests must still be routable into the private network. I am really using TH to hide the two ends of the call from each other, not to hide my internal network topology.

Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Date: Friday, July 29, 2016 at 8:40 AM
To: "Newlin, Ben" <Ben.Newlin at inin.com>, OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

Sorry to disagree, but IMHO they do exclude one each other.

Adding RR to TH should not be seen as a way of fixing some broken TH scenarios (with advertise).
So, let me try to understand what is not working for you. You do TH and advertise. In this case, normally, in the Contact headers generated by OpenSIPS (as a result of TH), it should be the TH interface, right ? What exactly seems to be the problem ? do you have a trace to show the issues ?

Regards,


Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 27.07.2016 16:05, Newlin, Ben wrote:
I understand that normally you would not need RR with TH, but the two concepts are not mutually exclusive in SIP. As I said, I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. This means that TH will populate the Contact header with the advertised address and if I cannot Record-Route with the actual address then I will not receive sequential requests.


Ben Newlin

From: Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Date: Wednesday, July 27, 2016 at 3:59 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>, "Newlin, Ben" <Ben.Newlin at inin.com><mailto:Ben.Newlin at inin.com>
Subject: Re: [OpenSIPS-Users] Record-Route and Dialog topology_hiding()

Hi Ben,

As I mentioned in different thread, TH is not compatible with the RR mechanism. If you do TH, your OpenSIPS will act as and end point (from SIP perspective), so there will be no Route/RR headers at all. So no need to do loose_route or so. You just do TH matching for the sequential requests and nothing more.

Regards,



Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

http://www.opensips-solutions.com
On 22.07.2016 16:48, Newlin, Ben wrote:
Hi,

I am using the Dialog module with topology_hiding() in my server and I have a need to Record-Route the call on my server as I am advertising a different address than I am listening on. I have found what I believe is an inconsistency in the handling of Record-Route within the Dialog topology_hiding functionality. The topology_hiding isn?t a true B2BUA, but it does set up different parameters for the incoming UAC and outgoing UAS sides of the call for the Via headers, Record-Route and Route headers, and the Contact header(s).

The problem is that the record_route() and loose_route() functions operate on different sides of the call. The record_route() function will only add a Record-Route header to the outgoing UAS side of the call. And since the record_route() function cannot be called from onreply_route, but is no way to add a Record-Route header to the UAC side of the call.

On the other hand, the loose_route() function only operates on the incoming UAC side of the call and there is no way to perform loose_route() on the UAS side of the call.

So there is a situation where Record-Route headers can only be added on the outgoing UAS side, but the associated Route headers can only be removed on the incoming UAC side (where they won?t exist since they can?t be added) and any added headers on the UAS side cannot be processed properly due to the lack of loose_route.

I can provide further information if this is unclear. It should be easily reproducible by attempting to use record_route in a topology_hiding scenario. The route is added to the outbound leg, but is not removed by loose_route so the message is looped back every time.

Ben Newlin | Sr Voice Network Engineer, PureCloud
phone & fax +1.317.957.1009 | ben.newlin at inin.com<mailto:ben.newlin at inin.com>
[age                       removed by sender.]
www.inin.com<http://www.inin.com>






_______________________________________________

Users mailing list

Users at lists.opensips.org<mailto:Users at lists.opensips.org>

http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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From Agalya_Ramachandran at comcast.com  Fri Jul 29 19:28:09 2016
From: Agalya_Ramachandran at comcast.com (Ramachandran, Agalya (Contractor))
Date: Fri, 29 Jul 2016 17:28:09 +0000
Subject: [OpenSIPS-Users] ACK method keeps looping
Message-ID: <11d8f9a85e284162910da715f29a2326@COPDCEX28.cable.comcast.com>

Hi team,

We are trying to run load test on OpenSIPS for our project.
We are using sipp tool to generate load test. We are facing some issues when we try to run a successful call.

Source SIPp ----------Opensips --------------->Dest SIPp

There are two scenarios.

1.       (Works) Failure code, send Invite from source SIPp and receive 480 .

2.       (Fails) Answering call, Send Invite from source SIPp  and receive 200 OK.
In first case, after receiving 480, OpenSIPS could able to send ACK to the destination SIPp
Where as in second case, there are two issues

         i.            After receiving 200 OK, OpenSIPS doesn't send ACK to destination SIPp. When it tries to send ACK out, it fails stating, usd_send_failed. Or

       ii.            Also the same ACK keeps looping somehow for huge number of times.

For the 2nd case, we are running only single call and it is failing.
Am attaching the log of SIPp messages and some snippet of OpenSIPS logs.

Have you come across this scenario? Why ACK is been looped so many times?
Also we see ACK message is so small, but it says, usd_send is failed due to message is long.
Any clue here would help us  a lot.

Regards,
Agalya



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From bogdan at opensips.org  Thu Jul 28 18:08:12 2016
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 28 Jul 2016 19:08:12 +0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <CAPu3kNVS6jV_weKfW_N1pqUszSGxAC_wvh07YMERCCC9=Y-RDA@mail.gmail.com>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
	<CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
	<CAPu3kNVS6jV_weKfW_N1pqUszSGxAC_wvh07YMERCCC9=Y-RDA@mail.gmail.com>
Message-ID: <579A2DEC.4090002@opensips.org>

Patrick,

if you just want to count them, do "opensipsctl fifo get_statistics 
active_dialogs"

See http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id296704

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 28.07.2016 16:38, Patrick Wakano wrote:
> From the linux CLI you can run:
> opensipsctl fifo dlg_list | grep -c "state:: 4"
>
> On Wed, Jul 27, 2016 at 4:57 PM, Cesar Alberto Rodriguez Fierro 
> <crf at transtelco.net <mailto:crf at transtelco.net>> wrote:
>
>     There was a wrong configuration of the "dialog.so" module.
>
>     I will try to parse the output in order to obtain the current calls.
>
>      Thanks for the help.
>
>     On Wed, Jul 27, 2016 at 1:20 PM, Carlos Eduardo <kaduww at gmail.com
>     <mailto:kaduww at gmail.com>> wrote:
>
>         Cesar,
>
>         Are you using the dialog module in your script?
>
>         This MI command will only return a valid value if the dialog
>         module is loaded (loadmodule "dialog.so") and if the dialogs
>         are criated durign the script processing (create_dialog()
>         command).
>
>         2016-07-27 16:12 GMT-03:00 Daniel Zanutti
>         <daniel.zanutti at gmail.com <mailto:daniel.zanutti at gmail.com>>:
>
>             On my sample, you should run:
>
>             opensipsctl fifo profile_get_size inbound
>
>
>             Using dlg_list you should get something like this:
>             # opensipsctl fifo dlg_list
>             dialog::  hash=84:1411689852
>                     state:: 4
>                     user_flags:: 0
>                     timestart:: 1469646635
>                     datestart:: 2016-07-27 16:10:35
>                     timeout:: 1469653835
>                     dateout:: 2016-07-27 18:10:35
>             ...
>             dialog::  hash=289:324429409
>                     state:: 2
>                     user_flags:: 0
>                     timestart:: 0
>                     timeout:: 0
>             ...
>             dialog::  hash=640:1695114669
>                     state:: 4
>             ...
>
>             Check if modules are successfully loaded.
>
>             Regards
>
>             On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez
>             Fierro <crf at transtelco.net <mailto:crf at transtelco.net>> wrote:
>
>
>                 Thanks for your help.
>
>                 I trying to use FIFO in order to  send requests to
>                 OpenSIPS, I have read some documentation about the
>                  Dialog Module.  I guess using the "opensipsctl fifo
>                 dlg_list" command can be useful to obtain the current
>                 calls, but I am not sure why the command is
>                 not available in my OpenSIPS version. When I execute
>                 the command ./opensipsctl fifo version, I am getting
>                 the following information
>                 Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>
>
>
>                 On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta
>                 Carvalho <pimenta at inatel.br
>                 <mailto:pimenta at inatel.br>> wrote:
>
>                     Now, thinking more about it, I would suggest you
>                     to put a SQL query in your proprietary software to
>                     query the OpenSIPS database directly.
>
>                     The table dialog will be always updated about
>                     current calls.
>
>
>                     Regards.
>
>
>                     RODRIGO PIMENTA CARVALHO
>                     Inatel Competence Center
>                     Software
>                     Ph: +55 35 3471 9200
>                     <tel:%2B55%2035%203471%209200> RAMAL 979
>
>
>                     ------------------------------------------------------------------------
>                     *De:* users-bounces at lists.opensips.org
>                     <mailto:users-bounces at lists.opensips.org>
>                     <users-bounces at lists.opensips.org
>                     <mailto:users-bounces at lists.opensips.org>> em nome
>                     de Rodrigo Pimenta Carvalho <pimenta at inatel.br
>                     <mailto:pimenta at inatel.br>>
>                     *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>                     *Para:* users at lists.opensips.org
>                     <mailto:users at lists.opensips.org>
>                     *Assunto:* Re: [OpenSIPS-Users] Get concurrent
>                     calls from sip server.
>
>                     With FIFO you can send requests to OpenSIPS, for
>                     example from a proprietary software. So, if a
>                     request wants to execute a query with avpops, I
>                     guess FIFO will be useful.
>
>
>                     Regards.
>
>
>
>                     RODRIGO PIMENTA CARVALHO
>                     Inatel Competence Center
>                     Software
>                     Ph: +55 35 3471 9200
>                     <tel:%2B55%2035%203471%209200> RAMAL 979
>
>
>                     ------------------------------------------------------------------------
>                     *De:* users-bounces at lists.opensips.org
>                     <mailto:users-bounces at lists.opensips.org>
>                     <users-bounces at lists.opensips.org
>                     <mailto:users-bounces at lists.opensips.org>> em nome
>                     de Cesar Alberto Rodriguez Fierro
>                     <crf at transtelco.net <mailto:crf at transtelco.net>>
>                     *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>                     *Para:* users at lists.opensips.org
>                     <mailto:users at lists.opensips.org>
>                     *Assunto:* [OpenSIPS-Users] Get concurrent calls
>                     from sip server.
>                     Hi !
>
>                     I am currently working in a project related with
>                     display in real time the active calls of our VoIP
>                     traffic, I would like to get the active sip-calls
>                     from a Kamailio Sip Server (running opensips), is
>                     there any way to obtain this information.
>
>                     Best Regards.
>
>
>
>
>                     Inline image 1|Cesar Rodriguez | VoiceOPS  | MX:
>                     +52 656-257-4112 <tel:%2B52%20656-257-4112> |
>
>                     CONFIDENTIALITY NOTICE:  This communication is
>                     intended only for the use of the individual or
>                     entity to which it is addressed and may contain
>                     information that is privileged, confidential, and
>                     exempt from disclosure under applicable law.  If
>                     you are not the intended recipient of this
>                     information, you are notified that any use,
>                     dissemination, distribution, or copying of the
>                     communication is strictly prohibited.
>
>                     _______________________________________________
>                     Users mailing list
>                     Users at lists.opensips.org
>                     <mailto:Users at lists.opensips.org>
>                     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>                 _______________________________________________
>                 Users mailing list
>                 Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>             _______________________________________________
>             Users mailing list
>             Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
>         -- 
>         *Carlos E. Wagner*
>         /Tecn?logo em Telecomunica??es, OCP, dCAA/
>         *
>         *
>         *Gnotel Telecom*
>         *E-mail:*_kaduww at gmail.com <mailto:kaduww at gmail.com>_
>         _carlos at gnotel.com.br <mailto:carlos at gnotel.com.br>_
>         *Fone:*+55 48 9981-0894 <tel:%2B55%2048%209981-0894>
>         *Skype:*carlos.e.wagner
>         www.gnotel.com.br <http://www.gnotel.com.br>
>         www.blogdovoip.com <http://www.blogdovoip.com>
>
>         _______________________________________________
>         Users mailing list
>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From pwakano at gmail.com  Thu Jul 28 19:21:15 2016
From: pwakano at gmail.com (Patrick Wakano)
Date: Thu, 28 Jul 2016 14:21:15 -0300
Subject: [OpenSIPS-Users] Get concurrent calls from sip server.
In-Reply-To: <579A2DEC.4090002@opensips.org>
References: <CAFgFmXq5DyjTGS_7EM6zYBGt9KgwWokfAEQYMPjVzeThOowhdw@mail.gmail.com>
	<BY2PR02MB1655B953FA1BE6B2C2533318B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<BY2PR02MB16556B29E3C341574A9AFD74B50F0@BY2PR02MB1655.namprd02.prod.outlook.com>
	<CAFgFmXodd1BP5xYqFPtvZy-r_LZB7fQuffVBu1v0uLecRnT0cw@mail.gmail.com>
	<CAN0jgiPNAhnx03qNjx4exB0rXVygVU+YmO19oonxKPqw=OyCtg@mail.gmail.com>
	<CAB=EceaC40KSK5keDnHCG63tPjK3w-vOf+AMqqA4hcXxuJ53kA@mail.gmail.com>
	<CAFgFmXo8dk_cA=7DJwhvuXei-mQbqOs1ZPSj4SN+HscMW-qSKQ@mail.gmail.com>
	<CAPu3kNVS6jV_weKfW_N1pqUszSGxAC_wvh07YMERCCC9=Y-RDA@mail.gmail.com>
	<579A2DEC.4090002@opensips.org>
Message-ID: <CAPu3kNWQB5tQf8Ae0QdBJ3TighP3N4-yRe+1BecY8xZCYXy-VA@mail.gmail.com>

Nice!
Thanks Bogdan!

On Thu, Jul 28, 2016 at 1:08 PM, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Patrick,
>
> if you just want to count them, do "opensipsctl fifo get_statistics
> active_dialogs"
>
> See http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id296704
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 28.07.2016 16:38, Patrick Wakano wrote:
>
> From the linux CLI you can run:
> opensipsctl fifo dlg_list | grep -c "state:: 4"
>
> On Wed, Jul 27, 2016 at 4:57 PM, Cesar Alberto Rodriguez Fierro <
> <crf at transtelco.net>crf at transtelco.net> wrote:
>
>> There was a wrong configuration of the  "dialog.so" module.
>>
>> I will try to parse the output in order to obtain the current calls.
>>
>>  Thanks for the help.
>>
>> On Wed, Jul 27, 2016 at 1:20 PM, Carlos Eduardo < <kaduww at gmail.com>
>> kaduww at gmail.com> wrote:
>>
>>> Cesar,
>>>
>>> Are you using the dialog module in your script?
>>>
>>> This MI command will only return a valid value if the dialog module is
>>> loaded (loadmodule "dialog.so") and if the dialogs are criated durign the
>>> script processing (create_dialog() command).
>>>
>>> 2016-07-27 16:12 GMT-03:00 Daniel Zanutti < <daniel.zanutti at gmail.com>
>>> daniel.zanutti at gmail.com>:
>>>
>>>> On my sample, you should run:
>>>>
>>>> opensipsctl fifo profile_get_size inbound
>>>>
>>>>
>>>> Using dlg_list you should get something like this:
>>>> # opensipsctl fifo dlg_list
>>>> dialog::  hash=84:1411689852
>>>>         state:: 4
>>>>         user_flags:: 0
>>>>         timestart:: 1469646635
>>>>         datestart:: 2016-07-27 16:10:35
>>>>         timeout:: 1469653835
>>>>         dateout:: 2016-07-27 18:10:35
>>>> ...
>>>> dialog::  hash=289:324429409
>>>>         state:: 2
>>>>         user_flags:: 0
>>>>         timestart:: 0
>>>>         timeout:: 0
>>>> ...
>>>> dialog::  hash=640:1695114669
>>>>         state:: 4
>>>> ...
>>>>
>>>> Check if modules are successfully loaded.
>>>>
>>>> Regards
>>>>
>>>> On Wed, Jul 27, 2016 at 3:41 PM, Cesar Alberto Rodriguez Fierro <
>>>> <crf at transtelco.net>crf at transtelco.net> wrote:
>>>>
>>>>>
>>>>> Thanks for your help.
>>>>>
>>>>> I trying to use  FIFO in order to  send requests to OpenSIPS, I have
>>>>> read some documentation about the  Dialog Module.  I guess using the "opensipsctl
>>>>> fifo dlg_list" command can be useful to obtain the current calls, but I am
>>>>> not sure why the command is not available in my OpenSIPS version.   When I
>>>>> execute the command ./opensipsctl fifo version, I am getting the following
>>>>> information
>>>>> Server:: OpenSIPS (1.8.2-notls (x86_64/linux)).
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Jul 27, 2016 at 11:42 AM, Rodrigo Pimenta Carvalho <
>>>>> <pimenta at inatel.br>pimenta at inatel.br> wrote:
>>>>>
>>>>>> Now, thinking more about it, I would suggest you to put a SQL query
>>>>>> in your proprietary software to query the OpenSIPS database directly.
>>>>>>
>>>>>> The table dialog will be always updated about current calls.
>>>>>>
>>>>>>
>>>>>> Regards.
>>>>>>
>>>>>>
>>>>>> RODRIGO PIMENTA CARVALHO
>>>>>> Inatel Competence Center
>>>>>> Software
>>>>>> Ph: +55 35 3471 9200 <%2B55%2035%203471%209200> RAMAL 979
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>> *De:* <users-bounces at lists.opensips.org>
>>>>>> users-bounces at lists.opensips.org < <users-bounces at lists.opensips.org>
>>>>>> users-bounces at lists.opensips.org> em nome de Rodrigo Pimenta
>>>>>> Carvalho < <pimenta at inatel.br>pimenta at inatel.br>
>>>>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:39
>>>>>> *Para:* <users at lists.opensips.org>users at lists.opensips.org
>>>>>> *Assunto:* Re: [OpenSIPS-Users] Get concurrent calls from sip server.
>>>>>>
>>>>>>
>>>>>> With FIFO you can send requests to OpenSIPS, for example from a
>>>>>> proprietary software. So, if a request wants to execute a query with
>>>>>> avpops, I guess FIFO will be useful.
>>>>>>
>>>>>>
>>>>>> Regards.
>>>>>>
>>>>>>
>>>>>>
>>>>>> RODRIGO PIMENTA CARVALHO
>>>>>> Inatel Competence Center
>>>>>> Software
>>>>>> Ph: +55 35 3471 9200 <%2B55%2035%203471%209200> RAMAL 979
>>>>>>
>>>>>>
>>>>>> ------------------------------
>>>>>> *De:* <users-bounces at lists.opensips.org>
>>>>>> users-bounces at lists.opensips.org < <users-bounces at lists.opensips.org>
>>>>>> users-bounces at lists.opensips.org> em nome de Cesar Alberto Rodriguez
>>>>>> Fierro < <crf at transtelco.net>crf at transtelco.net>
>>>>>> *Enviado:* quarta-feira, 27 de julho de 2016 14:29
>>>>>> *Para:* <users at lists.opensips.org>users at lists.opensips.org
>>>>>> *Assunto:* [OpenSIPS-Users] Get concurrent calls from sip server.
>>>>>>
>>>>>> Hi !
>>>>>>
>>>>>> I am currently working in a project related with display in real time
>>>>>> the active calls of our VoIP traffic, I would like to get the active
>>>>>> sip-calls from a Kamailio Sip Server (running opensips), is there any way
>>>>>> to obtain this information.
>>>>>>
>>>>>> Best Regards.
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> [image: Inline image 1] |Cesar Rodriguez | VoiceOPS  | MX: +52
>>>>>> 656-257-4112 <%2B52%20656-257-4112> |
>>>>>>
>>>>>> CONFIDENTIALITY NOTICE:  This communication is intended only for the
>>>>>> use of the individual or entity to which it is addressed and may contain
>>>>>> information that is privileged, confidential, and exempt from disclosure
>>>>>> under applicable law.  If you are not the intended recipient of this
>>>>>> information, you are notified that any use, dissemination, distribution, or
>>>>>> copying of the communication is strictly prohibited.
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> <Users at lists.opensips.org>Users at lists.opensips.org
>>>>>> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>>
>>>
>>>
>>> --
>>> *Carlos E. Wagner*
>>> *Tecn?logo em Telecomunica??es, OCP, dCAA*
>>>
>>> *Gnotel Telecom*
>>> *E-mail:* * <kaduww at gmail.com>kaduww at gmail.com <kaduww at gmail.com>*
>>>             * <carlos at gnotel.com.br>carlos at gnotel.com.br
>>> <carlos at gnotel.com.br>*
>>> *Fone:* +55 48 9981-0894 <%2B55%2048%209981-0894>
>>> *Skype:* carlos.e.wagner
>>> <http://www.gnotel.com.br>www.gnotel.com.br
>>> <http://www.blogdovoip.com>www.blogdovoip.com
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
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