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Tue Dec 2 06:11:19 CET 2008
help
-----Original Message-----
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Sent: Friday, November 28, 2008 6:18 PM
To: users at lists.opensips.org
Subject: Users Digest, Vol 4, Issue 102
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Today's Topics:
1. Re: Mediaproxy cannot write request to opensips mi_datagram
socket (Giuseppe Roberti)
2. Re: Fixing the Contact Header for NAT (I?aki Baz Castillo)
3. Problem with presence server (silverwaver)
4. Re: Problem with presence server (silverwaver)
5. Re: Codec question (Giuseppe Roberti)
6. Re: Fixing the Contact Header for NAT (Juan Backson)
----------------------------------------------------------------------
Message: 1
Date: Fri, 28 Nov 2008 13:22:31 +0100
From: Giuseppe Roberti <jnod at jnod.org>
Subject: Re: [OpenSIPS-Users] Mediaproxy cannot write request to
opensips mi_datagram socket
To: Dan Pascu <dan at ag-projects.com>
Cc: users at lists.opensips.org
Message-ID: <492FE287.6040303 at jnod.org>
Content-Type: text/plain; charset=iso-8859-1
Dan Pascu wrote:
> On Friday 28 November 2008, Giuseppe Roberti wrote:
>> None of that, i have found the error.
>>
>> The dispatcher read the configuration parameter for opensips socket
AS
>> IS, so i have to write the path without the apex.
>> Now that i have found this misconfiguration error i come with another
>> error. The setup is the same from my first email.
>> Here the log from dispatcher: http://rafb.net/p/j2U4YN96.html
>>
>> Thanks for helping.
>
> <quote>
> cannot write request to /tmp/opensips_socket: Transport endpoint is
not
> connected
> </quote>
>
> That means that /tmp/opensips_socket exists, but none is listening on
it.
>
Yes but i know that opensips mi_datagram works.
So maybe it can be a problem of mediaproxy.
I have created a link for /tmp/opensips_socket (/tmp/opensips.sock)
because it is used by opensipsctl when in UNIXSOCK mode.
See the log below.
root at maje:/tmp# ls -l opensips*
prw-rw---- 1 root root 0 2008-11-28 13:18 opensips_fifo
lrwxrwxrwx 1 root root 15 2008-11-28 13:17 opensips.sock ->
opensips_socket
srw-rw-rw- 1 root root 0 2008-11-28 13:18 opensips_socket
root at maje:/tmp# CTLENGINE=UNIXSOCK opensipsctl unixsock which
get_statistics
reset_statistics
uptime
version
pwd
arg
which
ps
kill
debug
list_blacklists
t_uac_dlg
t_uac_cancel
t_hash
t_reply
dlg_list
dlg_list_ctx
dlg_end_dlg
profile_get_size
profile_list_dlgs
ul_rm
ul_rm_contact
ul_dump
ul_flush
ul_add
ul_show_contact
sip_trace
nh_enable_ping
nh_enable_rtpp
nh_show_rtpp
root at maje:/tmp#
Regards.
--
Giuseppe Roberti
<jnod at jnod.org>
------------------------------
Message: 2
Date: Fri, 28 Nov 2008 13:23:04 +0100
From: I?aki Baz Castillo <ibc at aliax.net>
Subject: Re: [OpenSIPS-Users] Fixing the Contact Header for NAT
To: users at lists.opensips.org
Message-ID: <200811281323.04189.ibc at aliax.net>
Content-Type: text/plain; charset="utf-8"
El Viernes, 28 de Noviembre de 2008, Juan Backson escribi?:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
>
192.168.1.100:10314;received=121.15.89.95;branch=z9hG4bK-d87543-822f892d
753
>fdb57-1--d87543-;rport=10152
> Record-Route: <sip:192.168.1.101;lr=on;ftag=795a0d32;nat=yes>
This Record-Route means that the UAC MUST send the ACK to address
192.168.1.101 adding a Route header with same content. Is it feasible
for the
UAC to send the ACK to that private address?
> Contact: <sip:mod_sofia at 233.32.345.5:5800;transport=udp>
This Contact means that, when the ACK arrives to the proxy, it must
forward it
to address 233.32.345.5:5800 (since proxy does loose_routing and deletes
top
Route).
--
I?aki Baz Castillo
------------------------------
Message: 3
Date: Fri, 28 Nov 2008 20:24:10 +0800
From: silverwaver <silverwaver at gmail.com>
Subject: [OpenSIPS-Users] Problem with presence server
To: users at lists.opensips.org
Message-ID:
<a9c1609d0811280424x1ecc8d29peb96982c77cc18aa at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hello, everyone.
I have installed the OpenIMSCore and it runs ok. I can make a call
between Alice and
Bob using uctimsclient. Next step I want to install OpenSIPS as a
Presence Server.
Now I have installed OpenSIPS on the same machine with OpenIMSCore.
I've modified
Makefile to suppport presence server. My opensips.cfg file is
enclosed. OpenSIPS
starts ok after I run "opensipsctl start". I've set up the iFC in the
HSS to forward
all presence related messages to the Presence Server.
But I fail to use uctimsclient with presence service.I got the
following message:
Status-Line: SIP/2.0 600 Busy everywhere - Forwarding to S-CSCF
failed.
By a wireshark trace, I find that the SUBSCRIBE and PUBLISH messages
are correctly
forwarded to the presence server. It looks like that the presence server
doesn't
work. I don't know where is the problem now and how to test whether
the PS is ok.
Thank in advance,
silverwave
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Message: 4
Date: Fri, 28 Nov 2008 20:29:11 +0800
From: silverwaver <silverwaver at gmail.com>
Subject: Re: [OpenSIPS-Users] Problem with presence server
To: users at lists.opensips.org
Message-ID:
<a9c1609d0811280429i6cf68063ie3799116cfa39fe at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Here is my opensips.cfg.
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Message: 5
Date: Fri, 28 Nov 2008 13:34:05 +0100
From: Giuseppe Roberti <jnod at jnod.org>
Subject: Re: [OpenSIPS-Users] Codec question
To: Tseveendorj Ochirlantuu <tseveendorj at gmail.com>
Cc: users at lists.opensips.org
Message-ID: <492FE53D.4040206 at jnod.org>
Content-Type: text/plain; charset=ISO-8859-1
Hi.
OpenSIPS is not a media proxy.
It understand only SIP protocol.
You need some RTP proxy for dealing with audio/video (and so codecs).
The only two solution i know are:
- rtpproxy http://rtpproxy.org/
- mediaproxy http://mediaproxy-ng.org/
Regards.
Tseveendorj Ochirlantuu wrote:
> Hello,
> What kind of codecs included in OpenSIPS? G.711mu, G711a ,G729 ...
>
> Thank you
>
> Sincerely,
> Tseveen.
>
>
>
>
------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Giuseppe Roberti
<jnod at jnod.org>
------------------------------
Message: 6
Date: Fri, 28 Nov 2008 20:37:40 +0800
From: "Juan Backson" <juanbackson at gmail.com>
Subject: Re: [OpenSIPS-Users] Fixing the Contact Header for NAT
To: users at lists.opensips.org
Message-ID:
<27c25bc40811280437t2d1d9e3fi447049974f61a82b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi Inaki
Thank you for taking a look at the problem. I get that it is the
Record-Route that is not pointing back to opensips's public address and
instead it is using opensips private address.
The only place that I specified the private address 192.168.1.101 is
when I
start up opensips with the command:
/usr/local/sbin/opensips -l 192.168.1.101:5060 -f opensips.cfg
If I changed that to /usr/local/sbin/opensips -l 233.32.345.5:5060 -f
opensips.cfg, I am getting error saying:
ERROR:core;udp_init: bind(5,0x76512c,16) on 233.32.345.5: Cannot assign
requested address.
Is this the place where my setup is having problem or is this something
else?
Thanks alot for all your kind help.
JB
On Fri, Nov 28, 2008 at 8:23 PM, I?aki Baz Castillo <ibc at aliax.net>
wrote:
> El Viernes, 28 de Noviembre de 2008, Juan Backson escribi?:
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> > 192.168.1.100:10314;received=121.15.89.95
> ;branch=z9hG4bK-d87543-822f892d753
> >fdb57-1--d87543-;rport=10152
> > Record-Route: <sip:192.168.1.101;lr=on;ftag=795a0d32;nat=yes>
>
> This Record-Route means that the UAC MUST send the ACK to address
> 192.168.1.101 adding a Route header with same content. Is it feasible
for
> the
> UAC to send the ACK to that private address?
>
>
> > Contact: <sip:mod_sofia at 233.32.345.5:5800;transport=udp>
>
> This Contact means that, when the ACK arrives to the proxy, it must
forward
> it
> to address 233.32.345.5:5800 (since proxy does loose_routing and
deletes
> top
> Route).
>
> --
> I?aki Baz Castillo
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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