From razvan at opensips.org  Mon Jan  4 09:00:40 2021
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 4 Jan 2021 11:00:40 +0200
Subject: [OpenSIPS-Users] Transparent TLS
In-Reply-To: <1896060108.8933534.1609376271358@mail.yahoo.com>
References: <1896060108.8933534.1609376271358.ref@mail.yahoo.com>
 <1896060108.8933534.1609376271358@mail.yahoo.com>
Message-ID: <63a5d442-0d24-ee91-8580-896b3c89cacf@opensips.org>

Hi, Yavari!

Happy new year!
No, this is not possible - OpenSIPS is only able to route packages based 
on SIP packets - if you create an end-to-end connection between the 
client and media servers, OpenSIPS will not be able to decrypt the 
packages to know where to send what. OpenSIPS (and the entire SIP stack, 
by specifications) is not connection oriented, so packets can't be 
routed based on a previously established connection, only by SIP headers.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 12/31/20 2:57 AM, H Yavari via Users wrote:
> Hi to all,
> 
> Happy holidays.
> 
> In a distributed scenario, is it possible to have a TLS transparent with 
> Opensips?
> I mean clients make TLS connection with the nodes behind the proxy 
> server/load balancer and next time they can connect to the other nodes 
> but TLS connection is end to end between client and media server (AS/FS 
> etc.).
> Please advise.
> 
> Regards,
> HYavari
> 
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From saurabhc at 3clogic.com  Mon Jan  4 11:53:12 2021
From: saurabhc at 3clogic.com (Saurabh Chopra)
Date: Mon, 4 Jan 2021 17:23:12 +0530
Subject: [OpenSIPS-Users] Quality Routing Module in Opensips_3.1
In-Reply-To: <CAAmwJvHJOU0G7pWv1y=V47VWonDNofMLy0Mefayho7h+hHkNBA@mail.gmail.com>
References: <CAAmwJvGz89Hx1OBiXL_TUrN40mNgVnW6NJxiM93FHuN2SdgjDw@mail.gmail.com>
 <5468FE15-6A50-486A-8061-886037E2B548@gmail.com>
 <CAAmwJvHJOU0G7pWv1y=V47VWonDNofMLy0Mefayho7h+hHkNBA@mail.gmail.com>
Message-ID: <CAAmwJvH_ZJgckxoU1yqBg6CWnohwJNZkOaaQDoDL-h6gCLUy4A@mail.gmail.com>

Hi Tony/Opensips Team,

Happy New Year,

I have tried to test with default values in my configuration file but no
luck.The call is still going to the first gateway i.e. 104.XX.XX.XX. If
possible could you please help us at configuration side, what parameters
should be allowed to test this Qrouting module. Below is the output
for opensips-cli -x mi qr_status:-

                            "Carrier": {
                                "CRID": "cr1",
                                "Gateways": [
                                    {
                                        "GWID": "gw1",
                                        "ASR": "-1.00/9",
                                        "CCR": "-1.00/9",
                                        "PDD": "-1.00/7",
                                        "AST": "-1.00/7",
                                        "ACD": "-1.00/7"
                                    },
                                    {
                                        "GWID": "gw2",
                                        "ASR": "-1.00/0",
                                        "CCR": "-1.00/0",
                                        "PDD": "-1.00/0",
                                        "AST": "-1.00/0",
                                        "ACD": "-1.00/0"
                                    }
                                ]


Best Regards
Saurabh Chopra
+918861979979


On Mon, Dec 21, 2020 at 4:18 PM Saurabh Chopra <saurabhc at 3clogic.com> wrote:

> Hi Tony/Opensips Team,
>
> Will test it with default values as per your suggestion and will post the
> result of statistics for each of the gateways.
>
>
> Best Regards
> Saurabh Chopra
> +918861979979
>
>
> On Sun, Dec 20, 2020 at 3:09 PM Tomi Hakkarainen <tpaivaa at gmail.com>
> wrote:
>
>> Hi,
>>
>> never used myself but as reading the doc and your config, here some of my
>> thoughts.
>>
>> I see you are setting min_samples to zero and My guess is that that way
>> they will stay healthy forever?
>> Maybe adjust the config of min_samples to something like default or 15
>> and look how it behaves...
>> also have you viewed what the statistics show while testing? ( opensips-cli
>> -x mi qr_status )
>> Would like to hear how it goes :)
>>
>> Tomi
>>
>> On 18. Dec 2020, at 15.03, Saurabh Chopra <saurabhc at 3clogic.com> wrote:
>>
>> 
>> Hi All,
>>
>> Kindly update me on the query raised on Qrouting.
>>
>> Best Regards
>> Saurabh Chopra
>> +918861979979
>>
>>
>> On Thu, Dec 17, 2020 at 3:43 PM Saurabh Chopra <saurabhc at 3clogic.com>
>> wrote:
>>
>>> Hi All,
>>>
>>> I want to test the new quality routing module, previously i have tested
>>> the dynamic routing and it works for me. But somehow, qrouting module is
>>> not running as per my expectation. My understanding is qrouting module
>>> helps us to choose a better gateway at run time as per statistics like
>>> ASR,PDD,AST etc. I took two asterisk gateways
>>> 1:- 162.243.XX.XXX
>>> 2:- 104.131.XXX.XXX
>>>
>>> I have deliberately given 15sec wait on 104.131.XXX.XXX asterisk after
>>> this it will send 200 OK response for the call. So as per qrouting module,
>>> AST statistics for 104.131.XXX.XXX gateway would somewhat be lower than
>>> this 162.243.XX.XXX.
>>>
>>> So,I am expecting the call should mostly be reached to 162.243.XX.XXX
>>> gateway instead of 104.131.XXX.XXX, but this is not happening as calls are
>>> reaching to 104.131.XXX.XXX gateway which has poor statistics i.e AST.
>>>
>>> *Configuration done at mysql is given below:-*
>>> mysql> select * from dr_rules;
>>>
>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
>>> sort_alg | sort_profile | attrs | description        |
>>>
>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>> |      1 | 1       |        |         |        0 |         |
>>> gw2=50,gw1=50 | Q        |            1 |       | XXX_gateway |
>>>
>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>> 1 row in set (0.00 sec)
>>>
>>> mysql> select * from dr_gateways;
>>>
>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>> | id | gwid | type | address              | strip | pri_prefix | attrs |
>>> probe_mode | state | socket | description     |
>>>
>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>> |  1 | gw1  |    3 | 162.243.XX.XXX:5080  |     0 |            | NULL  |
>>>          0 |     0 | NULL   | 0                |
>>> |  2 | gw2  |    3 | 104.131.XXX.XXX:5080 |     0 |            | NULL  |
>>>          0 |     0 | NULL   | testing gateway2 |
>>>
>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>>
>>>
>>> *Configuration for loading qrouting module in opensips script is below:-*
>>> loadmodule "qrouting.so"
>>> modparam("qrouting", "db_url", "mysql://root:cccl0g1c at localhost
>>> /opensips")
>>> modparam("qrouting", "algorithm", "best-dest-first")
>>> modparam("qrouting", "history_span", 5)
>>> modparam("qrouting", "table_name", "qr_profiles")
>>> modparam("qrouting", "min_samples_pdd", 0)
>>> modparam("qrouting", "min_samples_ast", 0)
>>>
>>> Kindly help so that i can test this module successfully. Waiting for
>>> prompt response
>>>
>>> Best Regards
>>> Saurabh Chopra
>>> +918861979979
>>>
>> _______________________________________________
>> 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 Rajesh.Govindaraj at ipc.com  Mon Jan  4 16:59:19 2021
From: Rajesh.Govindaraj at ipc.com (Govindaraj, Rajesh)
Date: Mon, 4 Jan 2021 16:59:19 +0000
Subject: [OpenSIPS-Users] Topology hiding for presence:
 NOTIFY/Subscription refresh not successfully matching topology hiding
In-Reply-To: <BLAPR15MB3938D7B38627EE5F21AF05D5E3D80@BLAPR15MB3938.namprd15.prod.outlook.com>
References: <BLAPR15MB3938D7B38627EE5F21AF05D5E3D80@BLAPR15MB3938.namprd15.prod.outlook.com>
Message-ID: <BLAPR15MB3938C47EF82172973DB6E3D2E3D20@BLAPR15MB3938.namprd15.prod.outlook.com>

I triaged this issue further, the root cause was the contact was modified at opensips presence server using fixed_nated_contact API call as it traverses a non sip aware load balancer on the way. Due to this the NOTIFY request URI had a TCP ephemeral port and check_self API call failed and topology hiding match logic was not getting triggered.

Had to force the port to method check_self to 5060 and the issue got resolved. Not sure if there is a cleaner fix for this issue. I think the fact a non sip aware load balancer is in the path is the root cause. Please suggest if there is a way to fix this without any code change.

Thanks,

From: Govindaraj, Rajesh
Sent: Tuesday, December 29, 2020 1:15 PM
To: users at lists.opensips.org
Subject: Topology hiding for presence: NOTIFY/Subscription refresh not successfully matching topology hiding

Hi,

I am facing issues with topology hiding implementation for presence which was necessitated as existing TCP connections have to be used at Presence server and couldn't achieve this with record route routing and having original contact of application server.
Thanks for all your time and help. I am sure I am missing something small but I spent hours searching and reading up on Internet and would solicit your expertise to resolve this.

Objective: TCP transport for presence.

Topology:   opensips presence server <----> opensips proxy <----> IPC's Application Server.

Approach:

Case i: Without topology hiding and using record route:

In this case opensips proxy was adding two record route one for itself with sip:<ip>;transport=tcp<sip:%3cip%3e;transport=tcp> and via header carried rport. Opensips presence server while sending NOTIFY was throwing as TCP error,
Read through forum and understood that initial tcp request has to be re-used. Studied if alias can be used and also experimented with force_tcp_alias, but no luck.

Case ii: With topology hiding, no record route, use new contact:

With this approach able to get back initial NOTIFY
NOTIFY sip:172.29.109.119:40968;transport=tcp;thinfo=VG8tbzAdIFskPyccJRwmBhBQY31mX2RBckxiT2FkblpgWnpPJBMyPScfPx03SSUFI2gjAyMcIB00XGVmZltjCXVBMwdlaCcGIA4zBCMEICA9AD4GJ0kxESN+YxUjAC8aYlEiJTMcaxgvByMHMDowUiMGM1lhFzlqOx4qBjQXaAs+RVQaNB95RWBPYWNgQWFXcVpiVmlmZFlg SIP/2.0

With thinfo in request URI. Contact header of opensips sip server is present.

Now as per docs, tried to do topology_hiding_match by calling topology_hiding_match(), get this response,

DID NOT found,

I tried to add DID_NONE but don't see any log in the syslog.

The NOTIFY with contact header of opensip sip server is sent to Application Server. Record_route is called on this NOTIFY and record route is added without DID param.

When the subscription response comes back, the sample request below,(having the contact of opensips presence server no thinfo from 200ok for subscribe) the topology_hiding_match fails and the request does not go out.
I tried to load dialog module, call create_dialog but I understand that for subscribe the dialog would not be created. Please correct me if I am wrong. I also read about route header being used in opensips 2.1 per this thread,
https://opensips.org/pipermail/users/2017-December/038606.html but this is not being used in opensips version 2.4.7.

Not sure what am I missing. Please advise.

10.204.182.27 - Server running opensips proxy and application server.

10.29.109.130 - Opensips presence server

NOTIFY sent to application server:

NOTIFY sip:10.204.182.27:5059;transport=tcp;thinfo=VG8sbzAdIFskPyccJRwmBhBQY31mX2RBckxiT2FkblpgWnpPJBMyPScfPx03SSUFI2gjAyMcIB00XGVmZltjCXVBMwdlaCcGIA4zBCMEICA9AD4GJ0kxESN+JhYxXiZDYgNrJT4BaxgvByMHMDowUiMGM1lnFCJrbVY1WiBANldFUyELIFVyRH5TY2d6XmhdbUZnW2ZjYl8- SIP/2.0
Record-Route: <sip:10.204.182.27;r2=on;lr>
Record-Route: <sip:10.204.182.27;transport=tcp;r2=on;lr>
Via: SIP/2.0/UDP 10.204.182.27:5060;branch=z9hG4bK354b.18876e240190157c6feb29c18068a57a.0;i=685d8901
Via: SIP/2.0/TCP 10.42.3.115:5060;received=172.29.109.130;branch=z9hG4bK354b.faa91951.0
To: <sip:9797 at 10.204.182.27>;tag=19867159
From: <sip:9797-1 at ipc11.com>;tag=ab40-e3d19262d5e041c285ec0e9b00967d4b
CSeq: 1 NOTIFY
Call-ID: wlss-29dc9ccc-3d09899a4a9e634d0256bdf3c2cf8f0b at 10.204.182.27<mailto:wlss-29dc9ccc-3d09899a4a9e634d0256bdf3c2cf8f0b at 10.204.182.27>
Max-Forwards: 69
Content-Length: 566
User-Agent: OpenSIPS (2.4.8 (x86_64/linux))
Event: presence
Contact: <sip:sa at 10.29.109.130:5060;transport=tcp>
Subscription-State: active;expires=300
Content-Type: application/pidf+xml

Refresh subscribe:

Received SUBSCRIBE sip:sa at 10.29.109.130:5060;transport=tcp SIP/2.0^M
Content-Length: 0^M
CSeq: 2 SUBSCRIBE^M
Expires: 300^M
Route: <sip:10.204.182.27;r2=on;lr>^M
Route: <sip:10.204.182.27;transport=tcp;r2=on;lr>^M
Contact: <sip:wlssuser at 10.204.182.27:5059;transport=udp;wlsscid=65243f65cf6;sipappsessionid=app-7sa7kf2qbv51;wlssfcid=sip-1xgt8c89zz6kk>^M
Call-ID: wlss-af3350b7-c077bdf207ff802e84fa32ed40d47aed at 10.204.182.27^M<mailto:wlss-af3350b7-c077bdf207ff802e84fa32ed40d47aed at 10.204.182.27%5eM>
Max-Forwards: 70^M
From: <sip:9797 at 10.204.182.27>;tag=3427a3ff^M
To: <sip:9797-1 at ipc11.com>;tag=ab40-f97bec0eac0c0e4c851f049586838577^M
Event: presence^M
Via: SIP/2.0/UDP 10.204.182.27:5059;wlsscid=65243f65cf6;branch=z9hG4bK186c9181e96cf3053271dcd2b59330cd

Thanks,




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From bogdan at opensips.org  Tue Jan  5 11:43:22 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 5 Jan 2021 13:43:22 +0200
Subject: [OpenSIPS-Users] Introducing OpenSIPS 3.2
In-Reply-To: <365e8246-7634-e81c-8ff5-b02aa7246215@opensips.org>
References: <365e8246-7634-e81c-8ff5-b02aa7246215@opensips.org>
Message-ID: <648bbc2a-d09b-33b2-f802-2df2774f05e0@opensips.org>

A Happy New Year to you all !!

Have you filled in the poll for OpenSIPS 3.2 planning ? Time is ticking 
and the deadline is getting closer, so do it now https://bit.ly/2WDmAlV.

Keep in mind thatOpenSIPS is for the community, by the community. So 
your opinion matters!

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2020/

On 12/23/20 4:27 PM, Bogdan-Andrei Iancu wrote:
>
>
> Well, let's spin the wheel again for a new cycle – one more year, one 
> more evolution cycle, one more OpenSIPS major release. Even more, a 
> new topic is to be addressed. So let me introduce you to the upcoming 
> OpenSIPS 3.2 .
>
> For the upcoming OpenSIPS 3.2 release the main focus is on the 
> */in-cloud integration and distribution /*topic. Shortly said, this 
> translates into:
>
>   * distributed call center / queuing
>   * clustering support for modules
>   * Multi-level presence subscription
>   * RTP stream re-anchoring
>   * integration with Kafka, MQTT, Prometheus, ElasticSearch
>   * AWS support - DynamoDB, SSM, SQS, SNS
>   * script driven Back-2-Back
>
> For the full list with technical description and details, visit :
>
> https://www.opensips.org/Development/Opensips-3-2-Planning
>
>
> *IMPORTANT*
>
> As community is important to us and we want to align the OpenSIPS 
> roadmap with the needs of our users, be part of the shaping and 
> decision making for the OpenSIPS 3.2 Dev Plan via this *Feature Survey 
> <http://bit.ly/2WDmAlV>* - any feedback is important and it matters to us.
>
>
> Best regards and enjoy the winter holidays!!
>
> -- 
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>    https://opensips.org/training/OpenSIPS_eBootcamp_2020/

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From Sunil.More at samespace.com  Tue Jan  5 13:01:35 2021
From: Sunil.More at samespace.com (Sunil More)
Date: Tue, 5 Jan 2021 18:31:35 +0530
Subject: [OpenSIPS-Users] Opensips 3.1 and Homer 7
In-Reply-To: <CANWJJq8iJmsVAy5o17bKjnfbZ0UfRmifa+T1k6HO80M7w4Y9ug@mail.gmail.com>
References: <CAAP4-X=o-eizNHVW8oT9UmfSRrw-8dwFQMCe8NuqqWhyrKP7jQ@mail.gmail.com>
 <CAAP4-X=efmsNbznfisjA+REEuzh5adAo4E=r2cSvz-V2_FPJNg@mail.gmail.com>
 <CANWJJq8ffoGbLdZZ4W2bC=ywmoL+whY5J2odSmiH_qEiiDiNqA@mail.gmail.com>
 <CANWJJq8iJmsVAy5o17bKjnfbZ0UfRmifa+T1k6HO80M7w4Y9ug@mail.gmail.com>
Message-ID: <CAAP4-XmgN4n64-Xq2dhW=ENsWFB+qvAGfHr5x6hdbiDNingEdw@mail.gmail.com>

Hello All,

I am using homer 7 and opensips 3.1. I am using the tracer module along
with hep to send to helify-server.

here is a snippet of the config.

#----------------------- Loading Tracer
loadmodule "tracer.so"
modparam("tracer", "trace_on", 1)
modparam("tracer", "trace_id","[tid]uri=hep:hep_dst")

#----------------- Loading PROTOCOLs
-------------------------------------------------
loadmodule "proto_udp.so"
loadmodule "proto_tls.so"
loadmodule "proto_hep.so"
modparam("proto_hep", "hep_id", "[hep_dst] 10.153.53.167:9060
; transport=udp")
modparam("proto_hep", "homer5_on", 1)
modparam("proto_hep", "hep_port", 6666)

Using trace as below

trace("tid", "t", "sip|xlog",100);


I am able to get sip traces in Homer UI .  I am using rtpengine with
opensips and getting rtcp logs.  However

- I am not able to get Qos

- logs are sent to Homer but with *method RTP and type RTP.* (they are not
displayed in the logs tab)


I tried hep.js and I could populate Qos , SIP msgs and Logs in respective
tabs.


Could anyone please point me in the right direction?


Sunil More

Manager - DevOps

91 95033 38275

sunil.more at samespace.com



<https://www.samespace.com/>

-- 
 <http://www.samespace.com>



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From masked at vale.ski  Tue Jan  5 17:50:19 2021
From: masked at vale.ski (Michael Vale)
Date: Wed, 6 Jan 2021 04:50:19 +1100
Subject: [OpenSIPS-Users] www_challenge qop type failed with AAA
Message-ID: <E03414AB-518C-4CF7-93E4-0923EFB587BC@vale.ski>

Hi,

I’m attempting to do auth with AAA,

I was met with an error of no Cisco vendor and no SIP-URI-Host Attribute which I added to the dictionaries.

Now www_challenge thinks either “0” or “1” is an invalid qop type.										

										 ERROR:auth:fixup_qop: Bad qop type
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_cmd: Fixup failed for param [2]
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: Failed to fix command <www_challenge>
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: fixing failed (code=-1) at /etc/opensips/opensips.cfg:216
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:main: failed to fix configuration with err code -1
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: INFO:core:cleanup: cleanup
Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: NOTICE:core:main: Exiting....
Jan 06 04:42:14 unispy opensips[29709]: INFO:core:daemonize: pre-daemon proce

     if (!aaa_www_authorize("")) {
            www_challenge("", "1");
         };


Could someone please shed some light on the subject?

Also it would be nice if there was another approach to handling the SIP-URI-Host and Cisco issue or at least some confirmation that my approach was the best way to deal with it.


Regards,

Michael.

From vladp at opensips.org  Tue Jan  5 18:26:50 2021
From: vladp at opensips.org (Vlad Patrascu)
Date: Tue, 5 Jan 2021 20:26:50 +0200
Subject: [OpenSIPS-Users] www_challenge qop type failed with AAA
In-Reply-To: <E03414AB-518C-4CF7-93E4-0923EFB587BC@vale.ski>
References: <E03414AB-518C-4CF7-93E4-0923EFB587BC@vale.ski>
Message-ID: <2dc8c2b8-1df1-2cda-9d6e-51ab65bb86f6@opensips.org>

Hi Michael,

Regarding the qop issue, you should pass for that parameter a string 
value of "auth", "auth-int", or both separated by ','.

Regards,

-- 
Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com

On 05.01.2021 19:50, Michael Vale via Users wrote:
> Hi,
>
> I’m attempting to do auth with AAA,
>
> I was met with an error of no Cisco vendor and no SIP-URI-Host Attribute which I added to the dictionaries.
>
> Now www_challenge thinks either “0” or “1” is an invalid qop type.										
>
> 										 ERROR:auth:fixup_qop: Bad qop type
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_cmd: Fixup failed for param [2]
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: Failed to fix command <www_challenge>
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: fixing failed (code=-1) at /etc/opensips/opensips.cfg:216
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:main: failed to fix configuration with err code -1
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: INFO:core:cleanup: cleanup
> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: NOTICE:core:main: Exiting....
> Jan 06 04:42:14 unispy opensips[29709]: INFO:core:daemonize: pre-daemon proce
>
>       if (!aaa_www_authorize("")) {
>              www_challenge("", "1");
>           };
>
>
> Could someone please shed some light on the subject?
>
> Also it would be nice if there was another approach to handling the SIP-URI-Host and Cisco issue or at least some confirmation that my approach was the best way to deal with it.
>
>
> Regards,
>
> Michael.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From masked at vale.ski  Tue Jan  5 18:32:45 2021
From: masked at vale.ski (Michael Vale)
Date: Wed, 6 Jan 2021 05:32:45 +1100
Subject: [OpenSIPS-Users] www_challenge qop type failed with AAA
In-Reply-To: <2dc8c2b8-1df1-2cda-9d6e-51ab65bb86f6@opensips.org>
References: <E03414AB-518C-4CF7-93E4-0923EFB587BC@vale.ski>
 <2dc8c2b8-1df1-2cda-9d6e-51ab65bb86f6@opensips.org>
Message-ID: <0235E4F6-C75B-4C96-A46D-4159A52CD7F2@vale.ski>

The script wouldn’t load if I specified a auth-int as well as auth.  I removed auth-int entirely and the script runs.

> On 6 Jan 2021, at 5:26 am, Vlad Patrascu <vladp at opensips.org> wrote:
> 
> Hi Michael,
> 
> Regarding the qop issue, you should pass for that parameter a string value of "auth", "auth-int", or both separated by ','.
> 
> Regards,
> 
> -- 
> Vlad Patrascu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> 
> On 05.01.2021 19:50, Michael Vale via Users wrote:
>> Hi,
>> 
>> I’m attempting to do auth with AAA,
>> 
>> I was met with an error of no Cisco vendor and no SIP-URI-Host Attribute which I added to the dictionaries.
>> 
>> Now www_challenge thinks either “0” or “1” is an invalid qop type.										
>> 
>> 										 ERROR:auth:fixup_qop: Bad qop type
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_cmd: Fixup failed for param [2]
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: Failed to fix command <www_challenge>
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_actions: fixing failed (code=-1) at /etc/opensips/opensips.cfg:216
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:main: failed to fix configuration with err code -1
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: INFO:core:cleanup: cleanup
>> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: NOTICE:core:main: Exiting....
>> Jan 06 04:42:14 unispy opensips[29709]: INFO:core:daemonize: pre-daemon proce
>> 
>>      if (!aaa_www_authorize("")) {
>>             www_challenge("", "1");
>>          };
>> 
>> 
>> Could someone please shed some light on the subject?
>> 
>> Also it would be nice if there was another approach to handling the SIP-URI-Host and Cisco issue or at least some confirmation that my approach was the best way to deal with it.
>> 
>> 
>> Regards,
>> 
>> Michael.
>> _______________________________________________
>> 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 Jan  6 15:31:10 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 6 Jan 2021 17:31:10 +0200
Subject: [OpenSIPS-Users] Opensips 3.1 and Homer 7
In-Reply-To: <CAAP4-XmgN4n64-Xq2dhW=ENsWFB+qvAGfHr5x6hdbiDNingEdw@mail.gmail.com>
References: <CAAP4-X=o-eizNHVW8oT9UmfSRrw-8dwFQMCe8NuqqWhyrKP7jQ@mail.gmail.com>
 <CAAP4-X=efmsNbznfisjA+REEuzh5adAo4E=r2cSvz-V2_FPJNg@mail.gmail.com>
 <CANWJJq8ffoGbLdZZ4W2bC=ywmoL+whY5J2odSmiH_qEiiDiNqA@mail.gmail.com>
 <CANWJJq8iJmsVAy5o17bKjnfbZ0UfRmifa+T1k6HO80M7w4Y9ug@mail.gmail.com>
 <CAAP4-XmgN4n64-Xq2dhW=ENsWFB+qvAGfHr5x6hdbiDNingEdw@mail.gmail.com>
Message-ID: <89124d71-a67b-be18-fd8e-58566c8fee69@opensips.org>

Hi Sunil,

What is the exact version of opensips you are using ? do an "opensips 
-V" and check for the revision number.
Also

1) OpenSIPS does not sent any QoS data as it does not have it. This is 
something for RTPengine to do

2) The type of the log records was fixed, so consider an update to 
latest 3.1 (3.1.1)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2020/

On 1/5/21 3:01 PM, Sunil More wrote:
>
>
> Hello All,
>
> I am using homer 7 and opensips 3.1. I am using the tracer module 
> along with hep to send to helify-server.
>
> here is a snippet of the config.
>
> #----------------------- Loading Tracer
> loadmodule "tracer.so"
> modparam("tracer", "trace_on", 1)
> modparam("tracer", "trace_id","[tid]uri=hep:hep_dst")
> #----------------- Loading PROTOCOLs -------------------------------------------------
> loadmodule "proto_udp.so"
> loadmodule "proto_tls.so"
> loadmodule "proto_hep.so"
> modparam("proto_hep", "hep_id", "[hep_dst] 10.153.53.167:9060 
> <http://10.153.53.167:9060>; transport=udp")
> modparam("proto_hep", "homer5_on", 1)
> modparam("proto_hep", "hep_port", 6666)
>
> Using trace as below
>
> trace("tid", "t", "sip|xlog",100);
>
> I am able to get sip traces in Homer UI . I am using rtpengine with 
> opensips and getting rtcp logs.  However
>
> - I am not able to get Qos
>
> - logs are sent to Homer but with *method RTP and type RTP.* (they are 
> not displayed in the logs tab)
>
>
> I tried hep.js and I could populate Qos , SIP msgs and Logs in 
> respective tabs.
>
>
> Could anyone please point me in the right direction?
>
>
> Sunil More
>
> Manager - DevOps
>
> 91 95033 38275
>
> sunil.more at samespace.com <mailto:sunil.more at samespace.com>
>
> <https://www.samespace.com/>
>
>
>
>
> <http://www.samespace.com>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From mark at allenclan.co.uk  Wed Jan  6 15:59:36 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Wed, 6 Jan 2021 15:59:36 +0000
Subject: [OpenSIPS-Users] Mediaproxy Relay start error - 'MediaRelayBase' is
	not defined
Message-ID: <CADaqb1v9yr=cRPSZz8p+Vqwr19z2a62RDfA4aa3=e56fEKRHaA@mail.gmail.com>

Hi all - not sure what I'm missing here...

I'm installing Mediaproxy onto our Debian Buster box which is also running
OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...

15:40:07 opensipsx media-relay[4983]: INFO     Starting MediaProxy Relay
4.0.4
15:40:07 opensipsx media-relay[4983]: INFO     Set resource limit for
maximum open file descriptors to 11000
15:40:07 opensipsx media-relay[4983]: CRITICAL Failed to create MediaProxy
Relay: name 'MediaRelayBase' is not defined
15:40:07 opensipsx media-relay[4983]: ERROR    Traceback (most recent call
last):#012ERROR      File "/usr/bin/media-relay", line 100, in
<module>#012ERROR        from mediaproxy.relay import MediaRelay#012ERROR
   File "/usr/lib/python3/dist-packages/mediaproxy/relay.py", line 290, in
<module>#012ERROR        class MediaRelay(MediaRelayBase):#012ERROR
 NameError: name 'MediaRelayBase' is not defined
15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Main process
exited, code=exited, status=1/FAILURE
15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Failed with result
'exit-code'.

I'm starting the relay with the command...

systemctl start mediaproxy-relay

...I installed Mediaproxy using the Debian package using the instructions
at http://mediaproxy.ag-projects.com/installation-guide/ and
https://github.com/AGProjects/mediaproxy. mediaproxy-dispatcher is starting
successfully.

In the /etc/mediaproxy/config.ini file - everything is left at the default
setting except for...

dispatchers = xxx.xxx.xxx.xxx
advertised_ip = xxx.xxx.xxx.xxx

Certificates are in place in /etc/mediaproxy/tls


Anybody got any ideas about where I've gone wrong????
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From masked at vale.ski  Wed Jan  6 16:56:15 2021
From: masked at vale.ski (bobsy)
Date: Thu, 7 Jan 2021 03:56:15 +1100
Subject: [OpenSIPS-Users] Digest Auth with LDAP/RADIUS
Message-ID: <72D8F09A-954E-422C-A9D3-F6E90AF45961@vale.ski>

Hello everyone,

I’m attempting to use digest auth on Freeradius with LDAP and plaintext userPassword’s.

When the radius server goes to auth the digest hashes don’t match up.

  authenticate {
(17) digest: A1 = bobsy:opensips.vale.ski:password
(17) digest: A2 = REGISTER:sip:opensips.vale.ski
H(A1) = 0342aafbaea975d9fde3c46f3f093993
H(A2) = b0605d01a41aac18c7f1a84c8ca1c4f5
(17) digest: KD = 0342aafbaea975d9fde3c46f3f093993:5ff5eaca000015917970591b0edf7c7c6bbd13698c0dd5e6:b0605d01a41aac18c7f1a84c8ca1c4f5
EXPECTED a8d6639edfd61ac7b1bb247f7832b8e5
RECEIVED a817470a4e1612532d167bed0354a88b
(17) digest: FAILED authentication
(17)     [digest] = reject
(17)   } # authenticate = reject
(17) Failed to authenticate the user

I have calculate_ha1 set to 1.

Any insight would be great.

And after this is resolved maybe someone can help me find out why the Kerberos module looks for “User-Password”.  I believe it should be looking for “Cleartext-Password” and that’s why Kerberos won’t work for me.

Regards,

Michael Vale.

From ag at ag-projects.com  Wed Jan  6 21:57:03 2021
From: ag at ag-projects.com (Adrian Georgescu)
Date: Wed, 6 Jan 2021 18:57:03 -0300
Subject: [OpenSIPS-Users] Mediaproxy Relay start error -
 'MediaRelayBase' is not defined
In-Reply-To: <CADaqb1v9yr=cRPSZz8p+Vqwr19z2a62RDfA4aa3=e56fEKRHaA@mail.gmail.com>
References: <CADaqb1v9yr=cRPSZz8p+Vqwr19z2a62RDfA4aa3=e56fEKRHaA@mail.gmail.com>
Message-ID: <E3271891-24DE-41F3-8398-FBC5D46566E1@ag-projects.com>

This was a bug.

You must update to the latest mediaproxy version:

sudo apt update 
sudo apt install mediaproxy-relay mediaproxy-common mediaproxy-dispatcher

Regards,
Adrian

> On 6 Jan 2021, at 12:59, Mark Allen <mark at allenclan.co.uk> wrote:
> 
> Hi all - not sure what I'm missing here... 
> 
> I'm installing Mediaproxy onto our Debian Buster box which is also running OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...
> 
> 15:40:07 opensipsx media-relay[4983]: INFO     Starting MediaProxy Relay 4.0.4
> 15:40:07 opensipsx media-relay[4983]: INFO     Set resource limit for maximum open file descriptors to 11000
> 15:40:07 opensipsx media-relay[4983]: CRITICAL Failed to create MediaProxy Relay: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx media-relay[4983]: ERROR    Traceback (most recent call last):#012ERROR      File "/usr/bin/media-relay", line 100, in <module>#012ERROR        from mediaproxy.relay import MediaRelay#012ERROR      File "/usr/lib/python3/dist-packages/mediaproxy/relay.py", line 290, in <module>#012ERROR        class MediaRelay(MediaRelayBase):#012ERROR    NameError: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Main process exited, code=exited, status=1/FAILURE
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Failed with result 'exit-code'.
> 
> I'm starting the relay with the command...
> 
> systemctl start mediaproxy-relay
> 
> ...I installed Mediaproxy using the Debian package using the instructions at http://mediaproxy.ag-projects.com/installation-guide/ <http://mediaproxy.ag-projects.com/installation-guide/> and https://github.com/AGProjects/mediaproxy <https://github.com/AGProjects/mediaproxy>. mediaproxy-dispatcher is starting successfully.
> 
> In the /etc/mediaproxy/config.ini file - everything is left at the default setting except for...
> 
> dispatchers = xxx.xxx.xxx.xxx
> advertised_ip = xxx.xxx.xxx.xxx
> 
> Certificates are in place in /etc/mediaproxy/tls
> 
> 
> Anybody got any ideas about where I've gone wrong????
> 
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From Sunil.More at samespace.com  Thu Jan  7 05:23:44 2021
From: Sunil.More at samespace.com (Sunil More)
Date: Thu, 7 Jan 2021 10:53:44 +0530
Subject: [OpenSIPS-Users] Opensips 3.1 and Homer 7
In-Reply-To: <89124d71-a67b-be18-fd8e-58566c8fee69@opensips.org>
References: <CAAP4-X=o-eizNHVW8oT9UmfSRrw-8dwFQMCe8NuqqWhyrKP7jQ@mail.gmail.com>
 <CAAP4-X=efmsNbznfisjA+REEuzh5adAo4E=r2cSvz-V2_FPJNg@mail.gmail.com>
 <CANWJJq8ffoGbLdZZ4W2bC=ywmoL+whY5J2odSmiH_qEiiDiNqA@mail.gmail.com>
 <CANWJJq8iJmsVAy5o17bKjnfbZ0UfRmifa+T1k6HO80M7w4Y9ug@mail.gmail.com>
 <CAAP4-XmgN4n64-Xq2dhW=ENsWFB+qvAGfHr5x6hdbiDNingEdw@mail.gmail.com>
 <89124d71-a67b-be18-fd8e-58566c8fee69@opensips.org>
Message-ID: <CAAP4-XkjPON+yP4btJoHC=xRt-R7mtPcqLr9HdJxvqX6gCu1Nw@mail.gmail.com>

Hello Bogdan-Andrei,

Looks like I had two versions of opensips installed in different
prefixes and the one used was opensips 3.0.0 .
I created a fresh setup with opensips-3.1.1 and yes the logs are available.
I will take up the rtpengine issue with their channel.

Thanks.

Sunil More

Manager - DevOps

91 95033 38275

sunil.more at samespace.com



<https://www.samespace.com/>


On Wed, Jan 6, 2021 at 9:01 PM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Sunil,
>
> What is the exact version of opensips you are using ? do an "opensips -V"
> and check for the revision number.
> Also
>
> 1) OpenSIPS does not sent any QoS data as it does not have it. This is
> something for RTPengine to do
>
> 2) The type of the log records was fixed, so consider an update to latest
> 3.1 (3.1.1)
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>   https://opensips.org/training/OpenSIPS_eBootcamp_2020/
>
> On 1/5/21 3:01 PM, Sunil More wrote:
>
>
> Hello All,
>
> I am using homer 7 and opensips 3.1. I am using the tracer module along
> with hep to send to helify-server.
>
> here is a snippet of the config.
>
> #----------------------- Loading Tracer
> loadmodule "tracer.so"
> modparam("tracer", "trace_on", 1)
> modparam("tracer", "trace_id","[tid]uri=hep:hep_dst")
>
> #----------------- Loading PROTOCOLs -------------------------------------------------
> loadmodule "proto_udp.so"
> loadmodule "proto_tls.so"
> loadmodule "proto_hep.so"
> modparam("proto_hep", "hep_id", "[hep_dst] 10.153.53.167:9060
> ; transport=udp")
> modparam("proto_hep", "homer5_on", 1)
> modparam("proto_hep", "hep_port", 6666)
>
> Using trace as below
>
> trace("tid", "t", "sip|xlog",100);
>
> I am able to get sip traces in Homer UI .  I am using rtpengine with
> opensips and getting rtcp logs.  However
>
> - I am not able to get Qos
>
> - logs are sent to Homer but with *method RTP and type RTP.* (they are
> not displayed in the logs tab)
>
>
> I tried hep.js and I could populate Qos , SIP msgs and Logs in respective
> tabs.
>
>
> Could anyone please point me in the right direction?
>
>
> Sunil More
>
> Manager - DevOps
>
> 91 95033 38275
>
> sunil.more at samespace.com
>
>
>
> <https://www.samespace.com/>
>
>
>
> <http://www.samespace.com>
>
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>

-- 
 <http://www.samespace.com>



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From mark at allenclan.co.uk  Thu Jan  7 08:30:16 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 7 Jan 2021 08:30:16 +0000
Subject: [OpenSIPS-Users] Mediaproxy Relay start error -
 'MediaRelayBase' is not defined
In-Reply-To: <E3271891-24DE-41F3-8398-FBC5D46566E1@ag-projects.com>
References: <CADaqb1v9yr=cRPSZz8p+Vqwr19z2a62RDfA4aa3=e56fEKRHaA@mail.gmail.com>
 <E3271891-24DE-41F3-8398-FBC5D46566E1@ag-projects.com>
Message-ID: <CADaqb1ukFjXUOk9qCcZFv2kGzz+ZXi2SzYc4pcg=P8Xwte7t8A@mail.gmail.com>

Hi Adrian - thanks for getting back to me

I'm getting an unexpected error on apt-get update using the following in
sources.list...

    # AG Projects software
    deb http://ag-projects.com/debian unstable main
    deb-src http://ag-projects.com/debian unstable main

...I get...

    Err:10 https://ag-projects.com/debian unstable Release
         404  Not Found [IP: 85.17.186.10 443]
    E: The repository 'http://ag-projects.com/debian unstable Release' no
longer has a Release file.

...I've checked with nmap and I can see port 443 on 85.17.186.10

I've set it to use 'buster' instead of 'unstable' ('buster' was previously
giving me a similar error)...

    # AG Projects software
    deb http://ag-projects.com/debian buster main
    deb-src http://ag-projects.com/debian buster  main

...and that seems to now be working as I get...

    Hit:5 https://ag-projects.com/debian buster InRelease

...and when I install packages I'm seeing...

    Get:1 https://ag-projects.com/debian buster/main amd64
mediaproxy-common amd64 4.0.5buster [63.1 kB]
    Get:2 https://ag-projects.com/debian buster/main amd64
mediaproxy-dispatcher all 4.0.5buster [18.1 kB]
    Get:3 https://ag-projects.com/debian buster/main amd64 mediaproxy-relay
all 4.0.5buster [18.6 kB]
    Fetched 99.8 kB in 0s (535 kB/s)

...so hopefully I'm good to go now.

Thanks for your help. I'll post a follow-up once I've got it up and running








On Wed, 6 Jan 2021 at 21:59, Adrian Georgescu <ag at ag-projects.com> wrote:

> This was a bug.
>
> You must update to the latest mediaproxy version:
>
> sudo apt update
> sudo apt install mediaproxy-relay mediaproxy-common mediaproxy-dispatcher
>
> Regards,
> Adrian
>
> On 6 Jan 2021, at 12:59, Mark Allen <mark at allenclan.co.uk> wrote:
>
> Hi all - not sure what I'm missing here...
>
> I'm installing Mediaproxy onto our Debian Buster box which is also running
> OpenSIPS 3.1 but when I start the relay I'm getting an error in syslog...
>
> 15:40:07 opensipsx media-relay[4983]: INFO     Starting MediaProxy Relay
> 4.0.4
> 15:40:07 opensipsx media-relay[4983]: INFO     Set resource limit for
> maximum open file descriptors to 11000
> 15:40:07 opensipsx media-relay[4983]: CRITICAL Failed to create MediaProxy
> Relay: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx media-relay[4983]: ERROR    Traceback (most recent call
> last):#012ERROR      File "/usr/bin/media-relay", line 100, in
> <module>#012ERROR        from mediaproxy.relay import MediaRelay#012ERROR
>    File "/usr/lib/python3/dist-packages/mediaproxy/relay.py", line 290, in
> <module>#012ERROR        class MediaRelay(MediaRelayBase):#012ERROR
>  NameError: name 'MediaRelayBase' is not defined
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Main process
> exited, code=exited, status=1/FAILURE
> 15:40:07 opensipsx systemd[1]: mediaproxy-relay.service: Failed with
> result 'exit-code'.
>
> I'm starting the relay with the command...
>
> systemctl start mediaproxy-relay
>
> ...I installed Mediaproxy using the Debian package using the instructions
> at http://mediaproxy.ag-projects.com/installation-guide/ and
> https://github.com/AGProjects/mediaproxy. mediaproxy-dispatcher is
> starting successfully.
>
> In the /etc/mediaproxy/config.ini file - everything is left at the default
> setting except for...
>
> dispatchers = xxx.xxx.xxx.xxx
> advertised_ip = xxx.xxx.xxx.xxx
>
> Certificates are in place in /etc/mediaproxy/tls
>
>
> Anybody got any ideas about where I've gone wrong????
>
>
>
> _______________________________________________
> 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 mark at allenclan.co.uk  Thu Jan  7 10:37:12 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 7 Jan 2021 10:37:12 +0000
Subject: [OpenSIPS-Users] Mediaproxy Relay start error -
 'MediaRelayBase' is not defined
In-Reply-To: <CADaqb1ukFjXUOk9qCcZFv2kGzz+ZXi2SzYc4pcg=P8Xwte7t8A@mail.gmail.com>
References: <CADaqb1v9yr=cRPSZz8p+Vqwr19z2a62RDfA4aa3=e56fEKRHaA@mail.gmail.com>
 <E3271891-24DE-41F3-8398-FBC5D46566E1@ag-projects.com>
 <CADaqb1ukFjXUOk9qCcZFv2kGzz+ZXi2SzYc4pcg=P8Xwte7t8A@mail.gmail.com>
Message-ID: <CADaqb1vXwgoTtZyWbEWFrnR2V_sHCtzBvc8Fy=zpeujJXvHSyg@mail.gmail.com>

Yep - the software loaded successfully, integrated with OpenSIPS and I can
see in the log file that it is trying to handle RTP traffic. Just need to
work out the correct configuration now! :)

Thanks for the help Adrian
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From bogdan at opensips.org  Thu Jan  7 11:47:50 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jan 2021 13:47:50 +0200
Subject: [OpenSIPS-Users] [Feature] Script driven B2B with OpenSIPS 3.2
Message-ID: <42087cd3-0cb0-955e-3d10-e71908ccda74@opensips.org>

Hi all,

The work on 3.2 is already ongoing, and one by one, the new features 
will emerge.

One of the first (actually a continuation from 3.1) is the script driven 
B2B - shortly said, instead of using the nasty and intricate XML scripts 
to drive the B2B logic, you can now directly use the OpenSIPS routing 
script for that. Simpler and more powerful.

For more, see 
https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/

Enjoy it,

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com



From saurabhc at 3clogic.com  Thu Jan  7 12:04:30 2021
From: saurabhc at 3clogic.com (Saurabh Chopra)
Date: Thu, 7 Jan 2021 17:34:30 +0530
Subject: [OpenSIPS-Users] Quality Routing Module in Opensips_3.1
In-Reply-To: <CAAmwJvH_ZJgckxoU1yqBg6CWnohwJNZkOaaQDoDL-h6gCLUy4A@mail.gmail.com>
References: <CAAmwJvGz89Hx1OBiXL_TUrN40mNgVnW6NJxiM93FHuN2SdgjDw@mail.gmail.com>
 <5468FE15-6A50-486A-8061-886037E2B548@gmail.com>
 <CAAmwJvHJOU0G7pWv1y=V47VWonDNofMLy0Mefayho7h+hHkNBA@mail.gmail.com>
 <CAAmwJvH_ZJgckxoU1yqBg6CWnohwJNZkOaaQDoDL-h6gCLUy4A@mail.gmail.com>
Message-ID: <CAAmwJvEg4Bn=Xc0iMGb7bZ66Bir_YSxt94_w7mkMonYbuuSTFQ@mail.gmail.com>

Hi Opensips Team,

Could you please provide an update on the above query on Qrouting.

Best Regards
Saurabh Chopra
+918861979979


On Mon, Jan 4, 2021 at 5:23 PM Saurabh Chopra <saurabhc at 3clogic.com> wrote:

> Hi Tony/Opensips Team,
>
> Happy New Year,
>
> I have tried to test with default values in my configuration file but no
> luck.The call is still going to the first gateway i.e. 104.XX.XX.XX. If
> possible could you please help us at configuration side, what parameters
> should be allowed to test this Qrouting module. Below is the output
> for opensips-cli -x mi qr_status:-
>
>                             "Carrier": {
>                                 "CRID": "cr1",
>                                 "Gateways": [
>                                     {
>                                         "GWID": "gw1",
>                                         "ASR": "-1.00/9",
>                                         "CCR": "-1.00/9",
>                                         "PDD": "-1.00/7",
>                                         "AST": "-1.00/7",
>                                         "ACD": "-1.00/7"
>                                     },
>                                     {
>                                         "GWID": "gw2",
>                                         "ASR": "-1.00/0",
>                                         "CCR": "-1.00/0",
>                                         "PDD": "-1.00/0",
>                                         "AST": "-1.00/0",
>                                         "ACD": "-1.00/0"
>                                     }
>                                 ]
>
>
> Best Regards
> Saurabh Chopra
> +918861979979
>
>
> On Mon, Dec 21, 2020 at 4:18 PM Saurabh Chopra <saurabhc at 3clogic.com>
> wrote:
>
>> Hi Tony/Opensips Team,
>>
>> Will test it with default values as per your suggestion and will post the
>> result of statistics for each of the gateways.
>>
>>
>> Best Regards
>> Saurabh Chopra
>> +918861979979
>>
>>
>> On Sun, Dec 20, 2020 at 3:09 PM Tomi Hakkarainen <tpaivaa at gmail.com>
>> wrote:
>>
>>> Hi,
>>>
>>> never used myself but as reading the doc and your config, here some of
>>> my thoughts.
>>>
>>> I see you are setting min_samples to zero and My guess is that that way
>>> they will stay healthy forever?
>>> Maybe adjust the config of min_samples to something like default or 15
>>> and look how it behaves...
>>> also have you viewed what the statistics show while testing? ( opensips-cli
>>> -x mi qr_status )
>>> Would like to hear how it goes :)
>>>
>>> Tomi
>>>
>>> On 18. Dec 2020, at 15.03, Saurabh Chopra <saurabhc at 3clogic.com> wrote:
>>>
>>> 
>>> Hi All,
>>>
>>> Kindly update me on the query raised on Qrouting.
>>>
>>> Best Regards
>>> Saurabh Chopra
>>> +918861979979
>>>
>>>
>>> On Thu, Dec 17, 2020 at 3:43 PM Saurabh Chopra <saurabhc at 3clogic.com>
>>> wrote:
>>>
>>>> Hi All,
>>>>
>>>> I want to test the new quality routing module, previously i have tested
>>>> the dynamic routing and it works for me. But somehow, qrouting module is
>>>> not running as per my expectation. My understanding is qrouting module
>>>> helps us to choose a better gateway at run time as per statistics like
>>>> ASR,PDD,AST etc. I took two asterisk gateways
>>>> 1:- 162.243.XX.XXX
>>>> 2:- 104.131.XXX.XXX
>>>>
>>>> I have deliberately given 15sec wait on 104.131.XXX.XXX asterisk after
>>>> this it will send 200 OK response for the call. So as per qrouting module,
>>>> AST statistics for 104.131.XXX.XXX gateway would somewhat be lower than
>>>> this 162.243.XX.XXX.
>>>>
>>>> So,I am expecting the call should mostly be reached to 162.243.XX.XXX
>>>> gateway instead of 104.131.XXX.XXX, but this is not happening as calls are
>>>> reaching to 104.131.XXX.XXX gateway which has poor statistics i.e AST.
>>>>
>>>> *Configuration done at mysql is given below:-*
>>>> mysql> select * from dr_rules;
>>>>
>>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>>> | ruleid | groupid | prefix | timerec | priority | routeid | gwlist |
>>>> sort_alg | sort_profile | attrs | description        |
>>>>
>>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>>> |      1 | 1       |        |         |        0 |         |
>>>> gw2=50,gw1=50 | Q        |            1 |       | XXX_gateway |
>>>>
>>>> +--------+---------+--------+---------+----------+---------+---------------+----------+--------------+-------+--------------------+
>>>> 1 row in set (0.00 sec)
>>>>
>>>> mysql> select * from dr_gateways;
>>>>
>>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>>> | id | gwid | type | address              | strip | pri_prefix | attrs
>>>> | probe_mode | state | socket | description     |
>>>>
>>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>>> |  1 | gw1  |    3 | 162.243.XX.XXX:5080  |     0 |            | NULL
>>>>  |          0 |     0 | NULL   | 0                |
>>>> |  2 | gw2  |    3 | 104.131.XXX.XXX:5080 |     0 |            | NULL
>>>>  |          0 |     0 | NULL   | testing gateway2 |
>>>>
>>>> +----+------+------+----------------------+-------+------------+-------+------------+-------+--------+------------------+
>>>>
>>>>
>>>> *Configuration for loading qrouting module in opensips script is
>>>> below:-*
>>>> loadmodule "qrouting.so"
>>>> modparam("qrouting", "db_url", "mysql://root:cccl0g1c at localhost
>>>> /opensips")
>>>> modparam("qrouting", "algorithm", "best-dest-first")
>>>> modparam("qrouting", "history_span", 5)
>>>> modparam("qrouting", "table_name", "qr_profiles")
>>>> modparam("qrouting", "min_samples_pdd", 0)
>>>> modparam("qrouting", "min_samples_ast", 0)
>>>>
>>>> Kindly help so that i can test this module successfully. Waiting for
>>>> prompt response
>>>>
>>>> Best Regards
>>>> Saurabh Chopra
>>>> +918861979979
>>>>
>>> _______________________________________________
>>> 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 mark at allenclan.co.uk  Thu Jan  7 12:12:37 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 7 Jan 2021 12:12:37 +0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
Message-ID: <CADaqb1s4Aznb6Wmp2Vpq+QbMT6wniMZJOdu-QNk3X=THofN9CQ@mail.gmail.com>

I wonder if anyone can help me with this? I am trying to configure
Mediaproxy to handle RTP traffic coming from outside our local network.
Here's the setup:

    UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk

IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 10000 to
65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a Virtual
IP managed by keepalived.
UAC is MizuDroid app running on my Android phone connected to my home
network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
to our office network.
Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
system

SIP conversation between UAC and Asterisk via OpenSIPS looks to be working
fine. Endpoints connect, exchange data, and hangup. The problem is with SDP
addressing (NAT problem) causing no audio either way, which is what I want
Mediaproxy to handle.

In opensips.cfg I'm passing control for calls arriving at IPA to
Mediaproxy...

    if (is_method("INVITE")) {
        if (!has_totag()) {
            if ($fd == "4x.xxx.xxx.xxx") {
                xlog("Passing control to Mediaproxy...");
                engage_media_proxy();
            }
        }
    }

In /etc/mediaproxy/config.ini all settings are defaults except for setting
dispatcher as IPB...

    dispatchers = 192.168.xxx.xxx

...and I've tried it with and without advertised_ip set to IPA...

    advertised_ip = 4x.xxx.xxx.xxx


I can see that Mediaproxy is taking control of calls as instructed and
making changes to SDP but it's not solving my audio problems. What am I
doing wrong????
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From mark at allenclan.co.uk  Thu Jan  7 12:54:52 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 7 Jan 2021 12:54:52 +0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
In-Reply-To: <CADaqb1s4Aznb6Wmp2Vpq+QbMT6wniMZJOdu-QNk3X=THofN9CQ@mail.gmail.com>
References: <CADaqb1s4Aznb6Wmp2Vpq+QbMT6wniMZJOdu-QNk3X=THofN9CQ@mail.gmail.com>
Message-ID: <CADaqb1u6stAKbTaQJ4ru=2txyTOjWqhfEvEfWOn-QrMLtrC3Vw@mail.gmail.com>

Sorry... should have added that OpenSIPS box is acting as mid-registrar

On Thu, 7 Jan 2021, 12:12 Mark Allen, <mark at allenclan.co.uk> wrote:

> I wonder if anyone can help me with this? I am trying to configure
> Mediaproxy to handle RTP traffic coming from outside our local network.
> Here's the setup:
>
>     UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk
>
> IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 10000
> to 65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a
> Virtual IP managed by keepalived.
> UAC is MizuDroid app running on my Android phone connected to my home
> network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
> to our office network.
> Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
> system
>
> SIP conversation between UAC and Asterisk via OpenSIPS looks to be working
> fine. Endpoints connect, exchange data, and hangup. The problem is with SDP
> addressing (NAT problem) causing no audio either way, which is what I want
> Mediaproxy to handle.
>
> In opensips.cfg I'm passing control for calls arriving at IPA to
> Mediaproxy...
>
>     if (is_method("INVITE")) {
>         if (!has_totag()) {
>             if ($fd == "4x.xxx.xxx.xxx") {
>                 xlog("Passing control to Mediaproxy...");
>                 engage_media_proxy();
>             }
>         }
>     }
>
> In /etc/mediaproxy/config.ini all settings are defaults except for setting
> dispatcher as IPB...
>
>     dispatchers = 192.168.xxx.xxx
>
> ...and I've tried it with and without advertised_ip set to IPA...
>
>     advertised_ip = 4x.xxx.xxx.xxx
>
>
> I can see that Mediaproxy is taking control of calls as instructed and
> making changes to SDP but it's not solving my audio problems. What am I
> doing wrong????
>
>
>
>
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From solarmon at one-n.co.uk  Thu Jan  7 16:33:08 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Thu, 7 Jan 2021 16:33:08 +0000
Subject: [OpenSIPS-Users] Reject unsupported codec?
Message-ID: <CAOXfywPk9nuLJKOHOvHVJ-Un6126B__8LBxEj2MAzmUkizZAcQ@mail.gmail.com>

Hi,

On opensips 2.4.x how would I best check what codec is being offered, and
reject the call if it ONLY offers a codec that is not supported by us. For
example, if we only want to support G.711 PCMA/PCMU.

Thank you.
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From bogdan at opensips.org  Thu Jan  7 16:35:16 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jan 2021 18:35:16 +0200
Subject: [OpenSIPS-Users] Digest Auth with LDAP/RADIUS
In-Reply-To: <72D8F09A-954E-422C-A9D3-F6E90AF45961@vale.ski>
References: <72D8F09A-954E-422C-A9D3-F6E90AF45961@vale.ski>
Message-ID: <96b505d3-2299-b3dd-ee4d-6ab144938d64@opensips.org>

Hi Michael,

What you can do is to grab some online digest auth calculator and to 
doublecheck the auth responses on each side (opensips and radius)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2020/

On 1/6/21 6:56 PM, bobsy via Users wrote:
> Hello everyone,
>
> I’m attempting to use digest auth on Freeradius with LDAP and plaintext userPassword’s.
>
> When the radius server goes to auth the digest hashes don’t match up.
>
>    authenticate {
> (17) digest: A1 = bobsy:opensips.vale.ski:password
> (17) digest: A2 = REGISTER:sip:opensips.vale.ski
> H(A1) = 0342aafbaea975d9fde3c46f3f093993
> H(A2) = b0605d01a41aac18c7f1a84c8ca1c4f5
> (17) digest: KD = 0342aafbaea975d9fde3c46f3f093993:5ff5eaca000015917970591b0edf7c7c6bbd13698c0dd5e6:b0605d01a41aac18c7f1a84c8ca1c4f5
> EXPECTED a8d6639edfd61ac7b1bb247f7832b8e5
> RECEIVED a817470a4e1612532d167bed0354a88b
> (17) digest: FAILED authentication
> (17)     [digest] = reject
> (17)   } # authenticate = reject
> (17) Failed to authenticate the user
>
> I have calculate_ha1 set to 1.
>
> Any insight would be great.
>
> And after this is resolved maybe someone can help me find out why the Kerberos module looks for “User-Password”.  I believe it should be looking for “Cleartext-Password” and that’s why Kerberos won’t work for me.
>
> Regards,
>
> Michael Vale.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Thu Jan  7 16:37:02 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 7 Jan 2021 18:37:02 +0200
Subject: [OpenSIPS-Users] Reject unsupported codec?
In-Reply-To: <CAOXfywPk9nuLJKOHOvHVJ-Un6126B__8LBxEj2MAzmUkizZAcQ@mail.gmail.com>
References: <CAOXfywPk9nuLJKOHOvHVJ-Un6126B__8LBxEj2MAzmUkizZAcQ@mail.gmail.com>
Message-ID: <df5b2a6d-d0ac-a9b1-e66b-e01b97b670d0@opensips.org>

Hi,

See the codec checking related functions here -> 
https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_codec_exists

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2020/

On 1/7/21 6:33 PM, solarmon wrote:
> Hi,
>
> On opensips 2.4.x how would I best check what codec is being offered, 
> and reject the call if it ONLY offers a codec that is not supported by 
> us. For example, if we only want to support G.711 PCMA/PCMU.
>
> Thank you.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From solarmon at one-n.co.uk  Thu Jan  7 17:14:00 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Thu, 7 Jan 2021 17:14:00 +0000
Subject: [OpenSIPS-Users] Reject unsupported codec?
In-Reply-To: <df5b2a6d-d0ac-a9b1-e66b-e01b97b670d0@opensips.org>
References: <CAOXfywPk9nuLJKOHOvHVJ-Un6126B__8LBxEj2MAzmUkizZAcQ@mail.gmail.com>
 <df5b2a6d-d0ac-a9b1-e66b-e01b97b670d0@opensips.org>
Message-ID: <CAOXfywOxpEiPzJ4knQd3rGzvGvj6Z1RfBFamTbfmLEBPv+RcLg@mail.gmail.com>

Hi Bogdan,

That pointed me in the right direction and I'm able to reject calls. I
ended up using codec_exists_re()

Thank you!

On Thu, 7 Jan 2021 at 16:37, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi,
>
> See the codec checking related functions here ->
> https://opensips.org/html/docs/modules/2.4.x/sipmsgops.html#func_codec_exists
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>   https://opensips.org/training/OpenSIPS_eBootcamp_2020/
>
> On 1/7/21 6:33 PM, solarmon wrote:
>
> Hi,
>
> On opensips 2.4.x how would I best check what codec is being offered, and
> reject the call if it ONLY offers a codec that is not supported by us. For
> example, if we only want to support G.711 PCMA/PCMU.
>
> Thank you.
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From liviu at opensips.org  Fri Jan  8 11:03:40 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Fri, 8 Jan 2021 13:03:40 +0200
Subject: [OpenSIPS-Users] Quality Routing Module in Opensips_3.1
In-Reply-To: <CAAmwJvEg4Bn=Xc0iMGb7bZ66Bir_YSxt94_w7mkMonYbuuSTFQ@mail.gmail.com>
References: <CAAmwJvGz89Hx1OBiXL_TUrN40mNgVnW6NJxiM93FHuN2SdgjDw@mail.gmail.com>
 <5468FE15-6A50-486A-8061-886037E2B548@gmail.com>
 <CAAmwJvHJOU0G7pWv1y=V47VWonDNofMLy0Mefayho7h+hHkNBA@mail.gmail.com>
 <CAAmwJvH_ZJgckxoU1yqBg6CWnohwJNZkOaaQDoDL-h6gCLUy4A@mail.gmail.com>
 <CAAmwJvEg4Bn=Xc0iMGb7bZ66Bir_YSxt94_w7mkMonYbuuSTFQ@mail.gmail.com>
Message-ID: <aed42352-f4e8-50fd-411c-1b62b0ac228c@opensips.org>

On 07.01.2021 14:04, Saurabh Chopra wrote:
> Hi Opensips Team,
>
> Could you please provide an update on the above query on Qrouting.

Hey, Saurabh!

During OpenSIPS Summit 2020, I made a live demo [1] where the audience 
controlled the balancing of calls through qrouting, and it worked great!

While your pasted data looks completely fine and qrouting should work as 
expected, I have to ask: what does your "qr_profiles" table look like?  
Have you provisioned some proper thresholds for profile number "1"?  
FYI, by default, the table uses a "-1" value (disabled) for the 
thresholds of each monitored statistic.

[1]: https://youtu.be/uHFOB-J8GIQ?t=9366

Kind regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com



From johan at democon.be  Fri Jan  8 15:08:06 2021
From: johan at democon.be (johan)
Date: Fri, 8 Jan 2021 16:08:06 +0100
Subject: [OpenSIPS-Users] call api
Message-ID: <a655cd3b-9d0a-555d-2455-27e40c0a57f3@democon.be>

Hello Razvan,


is there already an opensips.cfg example config available for using call
api ?


wkr,

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From hyavari at rocketmail.com  Sat Jan  9 07:00:56 2021
From: hyavari at rocketmail.com (H Yavari)
Date: Sat, 9 Jan 2021 07:00:56 +0000 (UTC)
Subject: [OpenSIPS-Users] Transparent TLS
In-Reply-To: <63a5d442-0d24-ee91-8580-896b3c89cacf@opensips.org>
References: <1896060108.8933534.1609376271358.ref@mail.yahoo.com>
 <1896060108.8933534.1609376271358@mail.yahoo.com>
 <63a5d442-0d24-ee91-8580-896b3c89cacf@opensips.org>
Message-ID: <2080749410.58494.1610175656757@mail.yahoo.com>

 Hi Razvan,
Thanks for reply. But let me describe the scenario better:Clients must have TLS connection and we have an OpenSIPS cluster on the front of Asterisk servers. So in this case, if client's connection with one SIP proxy node goes down, it should be re-establish with other node in cluster? or as all cluster nodes are using shared DB and they talk to each other via BIN, client connection remains? thanks.
Regards.


    On Monday, January 4, 2021, 01:04:34 AM PST, Răzvan Crainea <razvan at opensips.org> wrote:  
 
 Hi, Yavari!

Happy new year!
No, this is not possible - OpenSIPS is only able to route packages based 
on SIP packets - if you create an end-to-end connection between the 
client and media servers, OpenSIPS will not be able to decrypt the 
packages to know where to send what. OpenSIPS (and the entire SIP stack, 
by specifications) is not connection oriented, so packets can't be 
routed based on a previously established connection, only by SIP headers.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 12/31/20 2:57 AM, H Yavari via Users wrote:
> Hi to all,
> 
> Happy holidays.
> 
> In a distributed scenario, is it possible to have a TLS transparent with 
> Opensips?
> I mean clients make TLS connection with the nodes behind the proxy 
> server/load balancer and next time they can connect to the other nodes 
> but TLS connection is end to end between client and media server (AS/FS 
> etc.).
> Please advise.
> 
> Regards,
> HYavari
> 
> 
> 
> 
> _______________________________________________
> 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 saurabhc at 3clogic.com  Mon Jan 11 06:11:18 2021
From: saurabhc at 3clogic.com (Saurabh Chopra)
Date: Mon, 11 Jan 2021 11:41:18 +0530
Subject: [OpenSIPS-Users] Quality Routing Module in Opensips_3.1
In-Reply-To: <aed42352-f4e8-50fd-411c-1b62b0ac228c@opensips.org>
References: <CAAmwJvGz89Hx1OBiXL_TUrN40mNgVnW6NJxiM93FHuN2SdgjDw@mail.gmail.com>
 <5468FE15-6A50-486A-8061-886037E2B548@gmail.com>
 <CAAmwJvHJOU0G7pWv1y=V47VWonDNofMLy0Mefayho7h+hHkNBA@mail.gmail.com>
 <CAAmwJvH_ZJgckxoU1yqBg6CWnohwJNZkOaaQDoDL-h6gCLUy4A@mail.gmail.com>
 <CAAmwJvEg4Bn=Xc0iMGb7bZ66Bir_YSxt94_w7mkMonYbuuSTFQ@mail.gmail.com>
 <aed42352-f4e8-50fd-411c-1b62b0ac228c@opensips.org>
Message-ID: <CAAmwJvER2-t3MWtVQF6n+FhTpomEec=NkfN2jfn3jKyV1+ve5A@mail.gmail.com>

Thank you liviu, I will go through the link and update the results.


Best Regards
Saurabh Chopra
+918861979979


On Fri, Jan 8, 2021 at 4:35 PM Liviu Chircu <liviu at opensips.org> wrote:

> On 07.01.2021 14:04, Saurabh Chopra wrote:
> > Hi Opensips Team,
> >
> > Could you please provide an update on the above query on Qrouting.
>
> Hey, Saurabh!
>
> During OpenSIPS Summit 2020, I made a live demo [1] where the audience
> controlled the balancing of calls through qrouting, and it worked great!
>
> While your pasted data looks completely fine and qrouting should work as
> expected, I have to ask: what does your "qr_profiles" table look like?
> Have you provisioned some proper thresholds for profile number "1"?
> FYI, by default, the table uses a "-1" value (disabled) for the
> thresholds of each monitored statistic.
>
> [1]: https://youtu.be/uHFOB-J8GIQ?t=9366
>
> Kind regards,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
> _______________________________________________
> 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 Jan 11 07:45:40 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 11 Jan 2021 09:45:40 +0200
Subject: [OpenSIPS-Users] Introducing OpenSIPS 3.2
In-Reply-To: <648bbc2a-d09b-33b2-f802-2df2774f05e0@opensips.org>
References: <365e8246-7634-e81c-8ff5-b02aa7246215@opensips.org>
 <648bbc2a-d09b-33b2-f802-2df2774f05e0@opensips.org>
Message-ID: <91710567-35ac-0224-be65-d373e873b08c@opensips.org>

Heads up, today is the last day for collecting your opinions on OpenSIPS 
3.2 roadmap.

https://bit.ly/2WDmAlV

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS Bootcamp 2020 online
   https://opensips.org/training/OpenSIPS_eBootcamp_2020/

On 1/5/21 1:43 PM, Bogdan-Andrei Iancu wrote:
> A Happy New Year to you all !!
>
> Have you filled in the poll for OpenSIPS 3.2 planning ? Time is 
> ticking and the deadline is getting closer, so do it now 
> https://bit.ly/2WDmAlV.
>
> Keep in mind thatOpenSIPS is for the community, by the community. So 
> your opinion matters!
>
> Best regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>    https://opensips.org/training/OpenSIPS_eBootcamp_2020/
> On 12/23/20 4:27 PM, Bogdan-Andrei Iancu wrote:
>>
>>
>> Well, let's spin the wheel again for a new cycle – one more year, one 
>> more evolution cycle, one more OpenSIPS major release. Even more, a 
>> new topic is to be addressed. So let me introduce you to the upcoming 
>> OpenSIPS 3.2 .
>>
>> For the upcoming OpenSIPS 3.2 release the main focus is on the 
>> */in-cloud integration and distribution /*topic. Shortly said, this 
>> translates into:
>>
>>   * distributed call center / queuing
>>   * clustering support for modules
>>   * Multi-level presence subscription
>>   * RTP stream re-anchoring
>>   * integration with Kafka, MQTT, Prometheus, ElasticSearch
>>   * AWS support - DynamoDB, SSM, SQS, SNS
>>   * script driven Back-2-Back
>>
>> For the full list with technical description and details, visit :
>>
>> https://www.opensips.org/Development/Opensips-3-2-Planning
>>
>>
>> *IMPORTANT*
>>
>> As community is important to us and we want to align the OpenSIPS 
>> roadmap with the needs of our users, be part of the shaping and 
>> decision making for the OpenSIPS 3.2 Dev Plan via this *Feature 
>> Survey <http://bit.ly/2WDmAlV>* - any feedback is important and it 
>> matters to us.
>>
>>
>> Best regards and enjoy the winter holidays!!
>>
>> -- 
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>    https://www.opensips-solutions.com
>> OpenSIPS Bootcamp 2020 online
>>    https://opensips.org/training/OpenSIPS_eBootcamp_2020/
>
>
> _______________________________________________
> 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  Mon Jan 11 08:31:37 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 11 Jan 2021 09:31:37 +0100
Subject: [OpenSIPS-Users] Transparent TLS
In-Reply-To: <2080749410.58494.1610175656757@mail.yahoo.com>
References: <1896060108.8933534.1609376271358.ref@mail.yahoo.com>
 <1896060108.8933534.1609376271358@mail.yahoo.com>
 <63a5d442-0d24-ee91-8580-896b3c89cacf@opensips.org>
 <2080749410.58494.1610175656757@mail.yahoo.com>
Message-ID: <CALXCt0pUMrYSM2-_9HN_xRw5ki+2FpEPCTW_TtkD=_9BTt8=xQ@mail.gmail.com>

Hi Yavari,

On Sat, Jan 9, 2021 at 8:03 AM H Yavari via Users <users at lists.opensips.org>
wrote:

> Clients must have TLS connection and we have an OpenSIPS cluster on the
> front of Asterisk servers. So in this case, if client's connection with one
> SIP proxy node goes down, it should be re-establish with other node in
> cluster? or as all cluster nodes are using shared DB and they talk to each
> other via BIN, client connection remains? thanks.
>
>
I do not think there is a way to have TCP (TLS, WebRTC, etc) connection to
survive a server failover.

You may want to have the clients to re-connect (reregister and reinvite) in
case of failover.

Or, maybe clustering OpenSIPSs in active-active via anycast.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From razvan at opensips.org  Mon Jan 11 10:05:00 2021
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Mon, 11 Jan 2021 12:05:00 +0200
Subject: [OpenSIPS-Users] call api
In-Reply-To: <a655cd3b-9d0a-555d-2455-27e40c0a57f3@democon.be>
References: <a655cd3b-9d0a-555d-2455-27e40c0a57f3@democon.be>
Message-ID: <8a9c1da7-88b4-d542-85ee-7ae45e705048@opensips.org>

Hi, Johan!

Check out this repo[1].

[1] 
https://github.com/razvancrainea/opensips-summit-distributed-2020/blob/call-api/opensips.cfg

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 1/8/21 5:08 PM, johan wrote:
> Hello Razvan,
> 
> 
> is there already an opensips.cfg example config available for using call
> api ?
> 
> 
> wkr,
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From Johan at democon.be  Mon Jan 11 10:11:00 2021
From: Johan at democon.be (Johan De Clercq)
Date: Mon, 11 Jan 2021 11:11:00 +0100
Subject: [OpenSIPS-Users] Transparent TLS
In-Reply-To: <CALXCt0pUMrYSM2-_9HN_xRw5ki+2FpEPCTW_TtkD=_9BTt8=xQ@mail.gmail.com>
References: <1896060108.8933534.1609376271358.ref@mail.yahoo.com>
 <1896060108.8933534.1609376271358@mail.yahoo.com>
 <63a5d442-0d24-ee91-8580-896b3c89cacf@opensips.org>
 <2080749410.58494.1610175656757@mail.yahoo.com>
 <CALXCt0pUMrYSM2-_9HN_xRw5ki+2FpEPCTW_TtkD=_9BTt8=xQ@mail.gmail.com>
Message-ID: <CAEVWGm-6TLA6d1zjbgFRVT+QNTmJ7cCjUEJxo4-h=e=QQEsLXg@mail.gmail.com>

Anycast can in my opinion only work in IP6.

Op ma 11 jan. 2021 om 09:35 schreef Giovanni Maruzzelli <gmaruzz at gmail.com>:

> Hi Yavari,
>
> On Sat, Jan 9, 2021 at 8:03 AM H Yavari via Users <
> users at lists.opensips.org> wrote:
>
>> Clients must have TLS connection and we have an OpenSIPS cluster on the
>> front of Asterisk servers. So in this case, if client's connection with one
>> SIP proxy node goes down, it should be re-establish with other node in
>> cluster? or as all cluster nodes are using shared DB and they talk to each
>> other via BIN, client connection remains? thanks.
>>
>>
> I do not think there is a way to have TCP (TLS, WebRTC, etc) connection to
> survive a server failover.
>
> You may want to have the clients to re-connect (reregister and reinvite)
> in case of failover.
>
> Or, maybe clustering OpenSIPSs in active-active via anycast.
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _______________________________________________
> 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  Mon Jan 11 10:12:47 2021
From: Johan at democon.be (Johan De Clercq)
Date: Mon, 11 Jan 2021 11:12:47 +0100
Subject: [OpenSIPS-Users] call api
In-Reply-To: <8a9c1da7-88b4-d542-85ee-7ae45e705048@opensips.org>
References: <a655cd3b-9d0a-555d-2455-27e40c0a57f3@democon.be>
 <8a9c1da7-88b4-d542-85ee-7ae45e705048@opensips.org>
Message-ID: <CAEVWGm9A13C4ECHpFjBKSA4ZHNN1kwzrC2bpPjMakj33i-DnnQ@mail.gmail.com>

Thanks Razvan, that's exactly the info that  I was looking for.
call api is very interesting to create an ACD like opensips instance.

Op ma 11 jan. 2021 om 11:08 schreef Răzvan Crainea <razvan at opensips.org>:

> Hi, Johan!
>
> Check out this repo[1].
>
> [1]
>
> https://github.com/razvancrainea/opensips-summit-distributed-2020/blob/call-api/opensips.cfg
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 1/8/21 5:08 PM, johan wrote:
> > Hello Razvan,
> >
> >
> > is there already an opensips.cfg example config available for using call
> > api ?
> >
> >
> > wkr,
> >
> >
> > _______________________________________________
> > 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 vinayak.makwana at ecosmob.com  Mon Jan 11 17:10:53 2021
From: vinayak.makwana at ecosmob.com (Vinayak Makwana)
Date: Mon, 11 Jan 2021 22:40:53 +0530
Subject: [OpenSIPS-Users] cache_raw_query for fetching data from mongodb in
	opensips 2.4.x
Message-ID: <CAPHmyfwt9fr_j4yQjk_0Nng9qYHonkV_=ASHAook7V_5XTDKPQ@mail.gmail.com>

hello everyone ,
I am using opensips 2.4.x. i want to know about how to use cache_raw_query
for fetching data from mongodb? Because their is no syntax for fetch data
in module.
Please help me  what changes I need to make into the opensips.cfg file?

-- 
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From bogdan at opensips.org  Tue Jan 12 11:43:42 2021
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 12 Jan 2021 13:43:42 +0200
Subject: [OpenSIPS-Users] OpenSIPS Control Panel 8.3.1 was released
Message-ID: <a8f5b0ef-c4ef-9a4d-9a3f-facefc069c46@opensips.org>

Hi all,

I know many of you were really anxious, so here we have the release of 
OpenSIPS Control Panel 8.3.1, compatible with OpenSIPS 3.1 . So, at this 
point we have the full 3.1 suite in place .

The OCP 8.3.1 has the same framework/engine as OCP 8.3.0 / 8.2.4, but it 
is aligned to the specifics of OpenSIPS 3.1 version:

* the "Dynamic Routing" tool was update to fit to the DB changed 
(related to sort algorithms)

* the "Call Center" tool was extended to cope with the new features 
related to dissuading and wrapup time

* the "Load Balancer" tool got the new "Attribute" field

* in the "CDRviewer" tool, the CDR filter was improved in terms of 
user-experience; also massive CDR exports are now supported.


Where to check for more or to download it?
http://controlpanel.opensips.org


Download and enjoy it as it's freshly baked for you,

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com



From adierlam at ptgi-ics.com  Tue Jan 12 20:14:10 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Tue, 12 Jan 2021 15:14:10 -0500
Subject: [OpenSIPS-Users] sangoma issue ?
Message-ID: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>

Hello All,

Wondering if anyone can offer some guidance on an issue seemingly with a
newly setup Sangoma D-500 Transcoding card.

Issue:
When using transcoding, soon after 1 transcoded call completes, utimer task
messages until opensips is restarted
seems like it happens when fetch_dlg_value: looking for <SngTc>

Jan 12 15:05:26 [21578] DBG:tm:timer_routine: timer
routine:2,tl=0x7f9ce5fd3cc0 next=(nil), timeout=16

Jan 12 15:05:26 [21578] DBG:tm:wait_handler: removing 0x7f9ce5fd3c40 from
table

Jan 12 15:05:26 [21578] DBG:tm:delete_cell: delete transaction
0x7f9ce5fd3c40

Jan 12 15:05:26 [21578] DBG:tm:run_trans_callbacks: trans=0x7f9ce5fd3c40,
callback type 4096, id 3 entered

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: destroying dialog
0x7f9ce5fcdc50

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: dlg expired or not in list
- dlg 0x7f9ce5fcdc50 [57:1450509359] with clid
'3b72b48a67a1eec840e2e1db12f8b37f at x.x.x.x' and tags 'as6d2cf831'
'Yjg9IHv3tknz-KgDTUF.K-lleno6X-hI'

Jan 12 15:05:26 [21578] DBG:dialog:run_dlg_callbacks:
dialog=0x7f9ce5fcdc50, type=2048

Jan 12 15:05:26 [21578] DBG:dialog:fetch_dlg_value: looking for <SngTc>

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 100 ms ago (now 16950 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 200 ms ago (now 17050 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 300 ms ago (now 17150 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 400 ms ago (now 17250 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 490 ms ago (now 17340 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 590 ms ago (now 17440 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 690 ms ago (now 17540 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 790 ms ago (now 17640 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer>
already scheduled 890 ms ago (now 17740 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:timer_ticker: timer task <tm-timer>
already scheduled 990 ms ago (now 17740 ms), delayin



opensips-cli trap output ( not sure what am looking at here, in case helps
)

#1  0x00007f9cde9aee5f in sangoma_worker_loop (proc_no=<optimized out>) at
sngtc_proc.c:46

        req = {type = REQ_FREE_SESSION, response_fd = 78, sng_req =
{usr_priv = 0x7f9ce8c8f86b <asctime_r+139>, tag = 15, rtcp_enable = 5
'\005', a = {codec_

id = 32668, ms = 20, host_ip = 0, host_netmask = 2021, host_udp_port = 0},
b = {codec_id = 9217376, ms = 0, host_ip = 3905485451, host_netmask =
32668, host_

udp_port = 20}}, sng_reply = 0x7f9ce5fd2c78}

        rc = 0

        __FUNCTION__ = "sangoma_worker_loop"

#2  0x00000000005045c8 in start_module_procs () at sr_module.c:858

        m = 0x7f9ce7beafa8

        n = 0

        l = 0

        x = <optimized out>

        __FUNCTION__ = "start_module_procs"

#3  0x000000000041e3df in main_loop () at main.c:779

        startup_done = 0x0

        chd_rank = 0

        last_check = 0

        rc = <optimized out>

#4  main (argc=<optimized out>, argv=<optimized out>) at main.c:1479

        c = <optimized out>

        r = 0

        tmp = 0x1 <Address 0x1 out of bounds>

        tmp_len = <optimized out>

        port = <optimized out>

        proto = <optimized out>

        protos_no = <optimized out>

        options = 0x664908
"f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"

        ret = -1

        seed = 4061548656

        rfd = <optimized out>

        __FUNCTION__ = "main"

thanks
Andy
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From spanda at 3clogic.com  Wed Jan 13 05:48:06 2021
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 13 Jan 2021 11:18:06 +0530
Subject: [OpenSIPS-Users] Need some help while configuring opensips 3.1
	with homer 7 .
In-Reply-To: <CAKN+pHYSjone=u+YS9-ANwPEV5bgNsJn25dJAHWeqwr642xGzg@mail.gmail.com>
References: <CALf-Q46v+Nx5tf8XNTX-Ma9DFnqU_nUSqEx42W-0m_LSikBzmg@mail.gmail.com>
 <CAKN+pHYSjone=u+YS9-ANwPEV5bgNsJn25dJAHWeqwr642xGzg@mail.gmail.com>
Message-ID: <CALf-Q466y43TCNteqUJeX0FWeDncwCB08V0Qe9PhV_LK9F4VKA@mail.gmail.com>

Hi Lorenzo ,

Need some understanding about the exported function  correlate in proto_hep
module .

The documentation says through this we can correlate two calls .

correlate("hep_dst", "correlation-no-1",$var(cor1),"correlation-no-2",
$var(cor2));


My scenario .
user1 -- opensips --- Freeswitch --- opensips --- user2

Here freeswitch behaves like a B2B UA . So when user1 calls to user2 there
is 2 call-id in both leg .
I want to correlate both the calls in a single flow . Freeswitch shares an
UUID in both the leg to identify these two calls . But in homer side the
call-id is he session id .

How I will do this ?

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


On Thu, Dec 17, 2020 at 6:19 PM Lorenzo Mangani <lorenzo.mangani at gmail.com>
wrote:

> Hi Sasmita,
>
> Please open an issue on the homer repository and we'll gladly assist you
> or any OpenSIPS community member :)
>
> https://github.com/sipcapture/homer/issues
>
>
> Kind Regards,
>
> Lorenzo Mangani
>
> QXIP BV - Capture Engineering
> Amsterdam, The Netherlands
>
> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is
> for the sole use of the intended recipient(s) and may contain confidential or
> legally privileged information. Any unauthorized review, use, disclosure or
> distribution is prohibited. If you are not the intended recipient, please contact
> the sender by reply e-mail and destroy all copies of this original message.
>
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From conorjpower at hotmail.com  Wed Jan 13 07:16:32 2021
From: conorjpower at hotmail.com (Conor Power)
Date: Wed, 13 Jan 2021 07:16:32 +0000
Subject: [OpenSIPS-Users] OpenSIPS as simple SIP proxy
Message-ID: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>

Hi,
   Apologies for the noob question but I'm hoping someone can point me in the right direction.

I am trying to use OpenSIPS as a simple proxy to proxy all calls to another SIP endpoint and back again to the original client. The only role of the OpenSIPS server is to function as the proxy and it is for all requests.

I have OpenSIPS up and running and can see the requests coming inbound using ngrep but I've had no success proxying the requests.

I added a sethostport() call in the config file but really am not sure where or how it fits in.

If someone might point me to a simple config file that would be used for such a proxy setup, it would be greatly appreciated.

Conor
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From Johan at democon.be  Wed Jan 13 08:27:45 2021
From: Johan at democon.be (Johan De Clercq)
Date: Wed, 13 Jan 2021 09:27:45 +0100
Subject: [OpenSIPS-Users] OpenSIPS as simple SIP proxy
In-Reply-To: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>
References: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>
Message-ID: <CAEVWGm-zhzcCo2-+aUWRjhu251pWhbCNcdEwrviRUevgx0Q1hw@mail.gmail.com>

take the residential config and see where that gets you.

Op wo 13 jan. 2021 om 08:21 schreef Conor Power <conorjpower at hotmail.com>:

> Hi,
>    Apologies for the noob question but I'm hoping someone can point me in
> the right direction.
>
> I am trying to use OpenSIPS as a simple proxy to proxy all calls to
> another SIP endpoint and back again to the original client. The only role
> of the OpenSIPS server is to function as the proxy and it is for all
> requests.
>
> I have OpenSIPS up and running and can see the requests coming inbound
> using ngrep but I've had no success proxying the requests.
>
> I added a sethostport() call in the config file but really am not sure
> where or how it fits in.
>
> If someone might point me to a simple config file that would be used for
> such a proxy setup, it would be greatly appreciated.
>
> Conor
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From masked at vale.ski  Wed Jan 13 09:18:00 2021
From: masked at vale.ski (bobsy)
Date: Wed, 13 Jan 2021 20:18:00 +1100
Subject: [OpenSIPS-Users] Digest Auth with LDAP/RADIUS
In-Reply-To: <96b505d3-2299-b3dd-ee4d-6ab144938d64@opensips.org>
References: <72D8F09A-954E-422C-A9D3-F6E90AF45961@vale.ski>
 <96b505d3-2299-b3dd-ee4d-6ab144938d64@opensips.org>
Message-ID: <774971A7-E107-4D32-9F46-03EAAA5A49FB@vale.ski>

Thanks Bogdan that useful to know.   Turns out I just typed the password in wrong!

> On 8 Jan 2021, at 3:35 am, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
> 
> Hi Michael,
> 
> What you can do is to grab some online digest auth calculator and to doublecheck the auth responses on each side (opensips and radius)
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>  https://www.opensips-solutions.com
> OpenSIPS Bootcamp 2020 online
>  https://opensips.org/training/OpenSIPS_eBootcamp_2020/
> 
> On 1/6/21 6:56 PM, bobsy via Users wrote:
>> Hello everyone,
>> 
>> I’m attempting to use digest auth on Freeradius with LDAP and plaintext userPassword’s.
>> 
>> When the radius server goes to auth the digest hashes don’t match up.
>> 
>>   authenticate {
>> (17) digest: A1 = bobsy:opensips.vale.ski:password
>> (17) digest: A2 = REGISTER:sip:opensips.vale.ski
>> H(A1) = 0342aafbaea975d9fde3c46f3f093993
>> H(A2) = b0605d01a41aac18c7f1a84c8ca1c4f5
>> (17) digest: KD = 0342aafbaea975d9fde3c46f3f093993:5ff5eaca000015917970591b0edf7c7c6bbd13698c0dd5e6:b0605d01a41aac18c7f1a84c8ca1c4f5
>> EXPECTED a8d6639edfd61ac7b1bb247f7832b8e5
>> RECEIVED a817470a4e1612532d167bed0354a88b
>> (17) digest: FAILED authentication
>> (17)     [digest] = reject
>> (17)   } # authenticate = reject
>> (17) Failed to authenticate the user
>> 
>> I have calculate_ha1 set to 1.
>> 
>> Any insight would be great.
>> 
>> And after this is resolved maybe someone can help me find out why the Kerberos module looks for “User-Password”.  I believe it should be looking for “Cleartext-Password” and that’s why Kerberos won’t work for me.
>> 
>> Regards,
>> 
>> Michael Vale.
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 



From farmorg at gmail.com  Wed Jan 13 12:27:15 2021
From: farmorg at gmail.com (Mark Farmer)
Date: Wed, 13 Jan 2021 12:27:15 +0000
Subject: [OpenSIPS-Users] local_route & append_hf
Message-ID: <CAOvQDet7h9FhWxJ9+fCuKWVhfBsV7-UWy7v=2oGVg5ZvmGjwpg@mail.gmail.com>

Hi everyone

I have append_hf in my local_route to add the contact header in for MS
Teams but the OPTIONS message does not contain the contact header:

OPTIONS sip:sip.pstnhub.microsoft.com:5061 SIP/2.0
Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5061;branch=z9hG4bK4e45.15b9f883.0
To: sip:sip.pstnhub.microsoft.com:5061
From: <sip:Anonymous at localhost>;tag=81adef1f13e5d48c34554037e606a809-cdf9
CSeq: 14 OPTIONS
Call-ID: 5918f5195ed9f0cf-315525 at 10.150.50.110
Max-Forwards: 70
Content-Length: 0
User-Agent: OpenSIPS

if (is_method("OPTIONS") && ($(rd{s.index, $var(dst)}) != NULL))
                xlog("CUSTOM_LOG: local_route Matched OPTIONS to $rd");
                append_hf("Contact: <sip:my.domain.com:5061
;transport=tls>\r\n");
                trace("htid","t");

It seems to be matching just fine:

CUSTOM_LOG: local_route 5
CUSTOM_LOG: local_route Matched OPTIONS to sip3.pstnhub.microsoft.com

I checked the doc for sipmsgops and append_hf does NOT say that it can be
used in local_route.

Any ideas?

Many thanks
Mark.
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From Sunil.More at samespace.com  Wed Jan 13 13:24:18 2021
From: Sunil.More at samespace.com (Sunil More)
Date: Wed, 13 Jan 2021 18:54:18 +0530
Subject: [OpenSIPS-Users] xlog message sent to homer 7 via HEP has type as
	ERROR
Message-ID: <CAAP4-XkNOCEVn-DeGGAef7_QLFB7GZ0ggacQ9BaAeRrADTWtpA@mail.gmail.com>

Hello All,

I am using opensips 3.1.1 along with homer 7 and I could observe that logs
are going to Homer with type as ERROR.

Here's opensips -V
version: opensips 3.1.1 (x86_64/linux)
git revision: 229ec0793
main.c compiled on 10:46:42 Jan  7 2021 with gcc 7

Here are  global  params
#-------------------- Global Parameters ------------------------
log_level=3
log_stderror=no
log_facility=LOG_LOCAL3
....

here's the log line in script
xlog("This is a outbound Call $tt/$si/$rm/$ci/$fU/$rU ");

here's the HEP ngrep trace log

HEP3.....................................
_......
.
..=.......d.....
...d.....
..5......
..5......hERROR:This is a outbound Call <null>/
10.153.53.157/INVITE/3BOQOHazTu/abcde/16468107000......3BOQOHazTu

Is there a way to control the type of log or is there something wrong that
I am doing here. I have also tried to log using

  xlog("L_INFO","This is a outbound Call $tt/$si/$rm/$ci/$fU/$rU ");  but
that doesn't go to HEP not does it show up in tail syslogs.



Sunil More

Manager - DevOps

91 95033 38275

sunil.more at samespace.com



<https://www.samespace.com/>

-- 
 <http://www.samespace.com>



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From liviu at opensips.org  Wed Jan 13 14:23:42 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Wed, 13 Jan 2021 16:23:42 +0200
Subject: [OpenSIPS-Users] xlog message sent to homer 7 via HEP has type
 as ERROR
In-Reply-To: <CAAP4-XkNOCEVn-DeGGAef7_QLFB7GZ0ggacQ9BaAeRrADTWtpA@mail.gmail.com>
References: <CAAP4-XkNOCEVn-DeGGAef7_QLFB7GZ0ggacQ9BaAeRrADTWtpA@mail.gmail.com>
Message-ID: <81085add-60f5-b5d4-209b-314e7930cb71@opensips.org>

On 13.01.2021 15:24, Sunil More wrote:
> I am using opensips 3.1.1 along with homer 7 and I could observe that 
> logs are going to Homer with type as ERROR.
>
> Here's opensips -V
> version: opensips 3.1.1 (x86_64/linux)
> git revision: 229ec0793
> main.c compiled on 10:46:42 Jan  7 2021 with gcc 7
>
Hi,

This has been fixed on Dec 10th [1], so you have 3 options:

* pull latest 3.1 source code and rebuild OpenSIPS
* install nightly 3.1 packages
* wait until 3.1.2 release

[1]: https://github.com/OpenSIPS/opensips/commit/2212865f19d

Cheers,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com



From Sunil.More at samespace.com  Wed Jan 13 14:50:56 2021
From: Sunil.More at samespace.com (Sunil More)
Date: Wed, 13 Jan 2021 20:20:56 +0530
Subject: [OpenSIPS-Users] xlog message sent to homer 7 via HEP has type
	as ERROR
In-Reply-To: <81085add-60f5-b5d4-209b-314e7930cb71@opensips.org>
References: <CAAP4-XkNOCEVn-DeGGAef7_QLFB7GZ0ggacQ9BaAeRrADTWtpA@mail.gmail.com>
 <81085add-60f5-b5d4-209b-314e7930cb71@opensips.org>
Message-ID: <CAAP4-Xn46F5FYReXBzS38rdAG872BZ3wbfSGS0YkuipMXtXXMw@mail.gmail.com>

Thanks Liviu Chircu . I will do one of those .


On Wed, Jan 13, 2021, 7:55 PM Liviu Chircu <liviu at opensips.org> wrote:

> On 13.01.2021 15:24, Sunil More wrote:
> > I am using opensips 3.1.1 along with homer 7 and I could observe that
> > logs are going to Homer with type as ERROR.
> >
> > Here's opensips -V
> > version: opensips 3.1.1 (x86_64/linux)
> > git revision: 229ec0793
> > main.c compiled on 10:46:42 Jan  7 2021 with gcc 7
> >
> Hi,
>
> This has been fixed on Dec 10th [1], so you have 3 options:
>
> * pull latest 3.1 source code and rebuild OpenSIPS
> * install nightly 3.1 packages
> * wait until 3.1.2 release
>
> [1]: https://github.com/OpenSIPS/opensips/commit/2212865f19d
>
> Cheers,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 
 <http://www.samespace.com>



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From arsperger at gmail.com  Wed Jan 13 14:59:30 2021
From: arsperger at gmail.com (Arsen Semenov)
Date: Wed, 13 Jan 2021 19:59:30 +0500
Subject: [OpenSIPS-Users] local_route & append_hf
In-Reply-To: <CAOvQDet7h9FhWxJ9+fCuKWVhfBsV7-UWy7v=2oGVg5ZvmGjwpg@mail.gmail.com>
References: <CAOvQDet7h9FhWxJ9+fCuKWVhfBsV7-UWy7v=2oGVg5ZvmGjwpg@mail.gmail.com>
Message-ID: <CAC3hFAqz1DpzuSWyhWYpDQFF3rFgyahwpG8qe7-Sm6OvPuSm0A@mail.gmail.com>

Hi,

The msg you've catched with trace("htid","t") you see it - as it is in the
buffer, before the changes are applied. The diff (removed or appended
headers) is being applied right before the message goes to wires.
So actually your header should be there.

On Wed, Jan 13, 2021 at 5:30 PM Mark Farmer <farmorg at gmail.com> wrote:

> Hi everyone
>
> I have append_hf in my local_route to add the contact header in for MS
> Teams but the OPTIONS message does not contain the contact header:
>
> OPTIONS sip:sip.pstnhub.microsoft.com:5061 SIP/2.0
> Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5061;branch=z9hG4bK4e45.15b9f883.0
> To: sip:sip.pstnhub.microsoft.com:5061
> From: <sip:Anonymous at localhost>;tag=81adef1f13e5d48c34554037e606a809-cdf9
> CSeq: 14 OPTIONS
> Call-ID: 5918f5195ed9f0cf-315525 at 10.150.50.110
> Max-Forwards: 70
> Content-Length: 0
> User-Agent: OpenSIPS
>
> if (is_method("OPTIONS") && ($(rd{s.index, $var(dst)}) != NULL))
>                 xlog("CUSTOM_LOG: local_route Matched OPTIONS to $rd");
>                 append_hf("Contact: <sip:my.domain.com:5061
> ;transport=tls>\r\n");
>                 trace("htid","t");
>
> It seems to be matching just fine:
>
> CUSTOM_LOG: local_route 5
> CUSTOM_LOG: local_route Matched OPTIONS to sip3.pstnhub.microsoft.com
>
> I checked the doc for sipmsgops and append_hf does NOT say that it can be
> used in local_route.
>
> Any ideas?
>
> Many thanks
> Mark.
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Arsen Semenov
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From adierlam at ptgi-ics.com  Wed Jan 13 15:26:03 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Wed, 13 Jan 2021 10:26:03 -0500
Subject: [OpenSIPS-Users] sangoma issue ?
In-Reply-To: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>
References: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>
Message-ID: <CAKe=zzhgq2vSeqgJeRtzjtn9yRPvf2C+Xd=ez3KoQbhWchaQpQ@mail.gmail.com>

forgot to mention this from:

opensips -V
version: opensips 3.0.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, HP_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, sigio_rt, select.
git revision: d19b20e
main.c compiled on 14:39:30 Dec 21 2020 with gcc 4.8.5

FYI get the same results from 2.4, 3.0, and 3.1





On Tue, Jan 12, 2021 at 3:14 PM Andy Dierlam <adierlam at ptgi-ics.com> wrote:

> Hello All,
>
> Wondering if anyone can offer some guidance on an issue seemingly with a
> newly setup Sangoma D-500 Transcoding card.
>
> Issue:
> When using transcoding, soon after 1 transcoded call completes, utimer
> task messages until opensips is restarted
> seems like it happens when fetch_dlg_value: looking for <SngTc>
>
> Jan 12 15:05:26 [21578] DBG:tm:timer_routine: timer
> routine:2,tl=0x7f9ce5fd3cc0 next=(nil), timeout=16
>
> Jan 12 15:05:26 [21578] DBG:tm:wait_handler: removing 0x7f9ce5fd3c40 from
> table
>
> Jan 12 15:05:26 [21578] DBG:tm:delete_cell: delete transaction
> 0x7f9ce5fd3c40
>
> Jan 12 15:05:26 [21578] DBG:tm:run_trans_callbacks: trans=0x7f9ce5fd3c40,
> callback type 4096, id 3 entered
>
> Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: destroying dialog
> 0x7f9ce5fcdc50
>
> Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: dlg expired or not in list
> - dlg 0x7f9ce5fcdc50 [57:1450509359] with clid
> '3b72b48a67a1eec840e2e1db12f8b37f at x.x.x.x' and tags 'as6d2cf831'
> 'Yjg9IHv3tknz-KgDTUF.K-lleno6X-hI'
>
> Jan 12 15:05:26 [21578] DBG:dialog:run_dlg_callbacks:
> dialog=0x7f9ce5fcdc50, type=2048
>
> Jan 12 15:05:26 [21578] DBG:dialog:fetch_dlg_value: looking for <SngTc>
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 100 ms ago (now 16950 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 200 ms ago (now 17050 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 300 ms ago (now 17150 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 400 ms ago (now 17250 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 490 ms ago (now 17340 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 590 ms ago (now 17440 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 690 ms ago (now 17540 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 790 ms ago (now 17640 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 890 ms ago (now 17740 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:timer_ticker: timer task <tm-timer>
> already scheduled 990 ms ago (now 17740 ms), delayin
>
>
>
> opensips-cli trap output ( not sure what am looking at here, in case helps
> )
>
> #1  0x00007f9cde9aee5f in sangoma_worker_loop (proc_no=<optimized out>)
> at sngtc_proc.c:46
>
>         req = {type = REQ_FREE_SESSION, response_fd = 78, sng_req =
> {usr_priv = 0x7f9ce8c8f86b <asctime_r+139>, tag = 15, rtcp_enable = 5
> '\005', a = {codec_
>
> id = 32668, ms = 20, host_ip = 0, host_netmask = 2021, host_udp_port = 0},
> b = {codec_id = 9217376, ms = 0, host_ip = 3905485451, host_netmask =
> 32668, host_
>
> udp_port = 20}}, sng_reply = 0x7f9ce5fd2c78}
>
>         rc = 0
>
>         __FUNCTION__ = "sangoma_worker_loop"
>
> #2  0x00000000005045c8 in start_module_procs () at sr_module.c:858
>
>         m = 0x7f9ce7beafa8
>
>         n = 0
>
>         l = 0
>
>         x = <optimized out>
>
>         __FUNCTION__ = "start_module_procs"
>
> #3  0x000000000041e3df in main_loop () at main.c:779
>
>         startup_done = 0x0
>
>         chd_rank = 0
>
>         last_check = 0
>
>         rc = <optimized out>
>
> #4  main (argc=<optimized out>, argv=<optimized out>) at main.c:1479
>
>         c = <optimized out>
>
>         r = 0
>
>         tmp = 0x1 <Address 0x1 out of bounds>
>
>         tmp_len = <optimized out>
>
>         port = <optimized out>
>
>         proto = <optimized out>
>
>         protos_no = <optimized out>
>
>         options = 0x664908
> "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"
>
>         ret = -1
>
>         seed = 4061548656
>
>         rfd = <optimized out>
>
>         __FUNCTION__ = "main"
>
> thanks
> Andy
>
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From farmorg at gmail.com  Wed Jan 13 15:39:55 2021
From: farmorg at gmail.com (Mark Farmer)
Date: Wed, 13 Jan 2021 15:39:55 +0000
Subject: [OpenSIPS-Users] local_route & append_hf
In-Reply-To: <CAC3hFAqz1DpzuSWyhWYpDQFF3rFgyahwpG8qe7-Sm6OvPuSm0A@mail.gmail.com>
References: <CAOvQDet7h9FhWxJ9+fCuKWVhfBsV7-UWy7v=2oGVg5ZvmGjwpg@mail.gmail.com>
 <CAC3hFAqz1DpzuSWyhWYpDQFF3rFgyahwpG8qe7-Sm6OvPuSm0A@mail.gmail.com>
Message-ID: <CAOvQDevDCz4gx9auR7m1yuEsOguFm-xnzronVao-xyxqGzE5_w@mail.gmail.com>

Thanks Arsen, that makes sense.

The trace is there so that I can use sngrep which in this case was
misleading :)

Mark.


On Wed, 13 Jan 2021 at 15:02, Arsen Semenov <arsperger at gmail.com> wrote:

> Hi,
>
> The msg you've catched with trace("htid","t") you see it - as it is in the
> buffer, before the changes are applied. The diff (removed or appended
> headers) is being applied right before the message goes to wires.
> So actually your header should be there.
>
> On Wed, Jan 13, 2021 at 5:30 PM Mark Farmer <farmorg at gmail.com> wrote:
>
>> Hi everyone
>>
>> I have append_hf in my local_route to add the contact header in for MS
>> Teams but the OPTIONS message does not contain the contact header:
>>
>> OPTIONS sip:sip.pstnhub.microsoft.com:5061 SIP/2.0
>> Via: SIP/2.0/TLS xxx.xxx.xxx.xxx:5061;branch=z9hG4bK4e45.15b9f883.0
>> To: sip:sip.pstnhub.microsoft.com:5061
>> From: <sip:Anonymous at localhost>;tag=81adef1f13e5d48c34554037e606a809-cdf9
>> CSeq: 14 OPTIONS
>> Call-ID: 5918f5195ed9f0cf-315525 at 10.150.50.110
>> Max-Forwards: 70
>> Content-Length: 0
>> User-Agent: OpenSIPS
>>
>> if (is_method("OPTIONS") && ($(rd{s.index, $var(dst)}) != NULL))
>>                 xlog("CUSTOM_LOG: local_route Matched OPTIONS to $rd");
>>                 append_hf("Contact: <sip:my.domain.com:5061
>> ;transport=tls>\r\n");
>>                 trace("htid","t");
>>
>> It seems to be matching just fine:
>>
>> CUSTOM_LOG: local_route 5
>> CUSTOM_LOG: local_route Matched OPTIONS to sip3.pstnhub.microsoft.com
>>
>> I checked the doc for sipmsgops and append_hf does NOT say that it can be
>> used in local_route.
>>
>> Any ideas?
>>
>> Many thanks
>> Mark.
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Arsen Semenov
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Mark Farmer
farmorg at gmail.com
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From mark at allenclan.co.uk  Wed Jan 13 16:08:27 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Wed, 13 Jan 2021 16:08:27 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
Message-ID: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>

Hi all - I've been banging my head against this but not succeeding.

Our setup...

UAC           192.168.x.x
  |
Router        5.x.x.x
  |
(internet)
  |
Firewall      46.x.x.x maps
  |           directly to
OpenSIPS      192.168.x.x      Mid-registrar
  |
Asterisk      192.168.x.x


Current situation:
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk
box
- Destination extension rings and can pick up call
- There is no audio either way & call drops after about 30 secs (Asterisk
kills call with "Requested channel not available" because not RTP traffic
is reaching destination)

I have tried passing audio through Mediaproxy on OpenSIPS box but with no
success. Using Wireshark I can see RTP traffic initiated at both ends, but
it doesn't reach the other end either way.

Is there some definitive guide to setting this up correctly or are there
specific steps that I need to follow?
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From johan at democon.be  Wed Jan 13 16:43:39 2021
From: johan at democon.be (Johan De Clercq)
Date: Wed, 13 Jan 2021 16:43:39 +0000
Subject: [OpenSIPS-Users] sangoma issue ?
In-Reply-To: <CAKe=zzhgq2vSeqgJeRtzjtn9yRPvf2C+Xd=ez3KoQbhWchaQpQ@mail.gmail.com>
References: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>,
 <CAKe=zzhgq2vSeqgJeRtzjtn9yRPvf2C+Xd=ez3KoQbhWchaQpQ@mail.gmail.com>
Message-ID: <PR1PR06MB46837F7B3E7053F867CFAF49A7A90@PR1PR06MB4683.eurprd06.prod.outlook.com>

Why not use rtpengine to do transcoding?

Outlook voor iOS<https://aka.ms/o0ukef> downloaden
________________________________
Van: Users <users-bounces at lists.opensips.org> namens Andy Dierlam <adierlam at ptgi-ics.com>
Verzonden: Wednesday, January 13, 2021 4:26:03 PM
Aan: users at lists.opensips.org <users at lists.opensips.org>
Onderwerp: Re: [OpenSIPS-Users] sangoma issue ?

forgot to mention this from:

opensips -V
version: opensips 3.0.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_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, sigio_rt, select.
git revision: d19b20e
main.c compiled on 14:39:30 Dec 21 2020 with gcc 4.8.5

FYI get the same results from 2.4, 3.0, and 3.1





On Tue, Jan 12, 2021 at 3:14 PM Andy Dierlam <adierlam at ptgi-ics.com<mailto:adierlam at ptgi-ics.com>> wrote:
Hello All,

Wondering if anyone can offer some guidance on an issue seemingly with a newly setup Sangoma D-500 Transcoding card.

Issue:
When using transcoding, soon after 1 transcoded call completes, utimer task messages until opensips is restarted
seems like it happens when fetch_dlg_value: looking for <SngTc>


Jan 12 15:05:26 [21578] DBG:tm:timer_routine: timer routine:2,tl=0x7f9ce5fd3cc0 next=(nil), timeout=16

Jan 12 15:05:26 [21578] DBG:tm:wait_handler: removing 0x7f9ce5fd3c40 from table

Jan 12 15:05:26 [21578] DBG:tm:delete_cell: delete transaction 0x7f9ce5fd3c40

Jan 12 15:05:26 [21578] DBG:tm:run_trans_callbacks: trans=0x7f9ce5fd3c40, callback type 4096, id 3 entered

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: destroying dialog 0x7f9ce5fcdc50

Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f9ce5fcdc50 [57:1450509359] with clid '3b72b48a67a1eec840e2e1db12f8b37f at x.x.x.x' and tags 'as6d2cf831' 'Yjg9IHv3tknz-KgDTUF.K-lleno6X-hI'

Jan 12 15:05:26 [21578] DBG:dialog:run_dlg_callbacks: dialog=0x7f9ce5fcdc50, type=2048

Jan 12 15:05:26 [21578] DBG:dialog:fetch_dlg_value: looking for <SngTc>

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 100 ms ago (now 16950 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 200 ms ago (now 17050 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 300 ms ago (now 17150 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 400 ms ago (now 17250 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 490 ms ago (now 17340 ms), delaying execution

Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 590 ms ago (now 17440 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 690 ms ago (now 17540 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 790 ms ago (now 17640 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled 890 ms ago (now 17740 ms), delaying execution

Jan 12 15:05:27 [21561] WARNING:core:timer_ticker: timer task <tm-timer> already scheduled 990 ms ago (now 17740 ms), delayin



opensips-cli trap output ( not sure what am looking at here, in case helps )


#1  0x00007f9cde9aee5f in sangoma_worker_loop (proc_no=<optimized out>) at sngtc_proc.c:46

        req = {type = REQ_FREE_SESSION, response_fd = 78, sng_req = {usr_priv = 0x7f9ce8c8f86b <asctime_r+139>, tag = 15, rtcp_enable = 5 '\005', a = {codec_

id = 32668, ms = 20, host_ip = 0, host_netmask = 2021, host_udp_port = 0}, b = {codec_id = 9217376, ms = 0, host_ip = 3905485451, host_netmask = 32668, host_

udp_port = 20}}, sng_reply = 0x7f9ce5fd2c78}

        rc = 0

        __FUNCTION__ = "sangoma_worker_loop"

#2  0x00000000005045c8 in start_module_procs () at sr_module.c:858

        m = 0x7f9ce7beafa8

        n = 0

        l = 0

        x = <optimized out>

        __FUNCTION__ = "start_module_procs"

#3  0x000000000041e3df in main_loop () at main.c:779

        startup_done = 0x0

        chd_rank = 0

        last_check = 0

        rc = <optimized out>

#4  main (argc=<optimized out>, argv=<optimized out>) at main.c:1479

        c = <optimized out>

        r = 0

        tmp = 0x1 <Address 0x1 out of bounds>

        tmp_len = <optimized out>

        port = <optimized out>

        proto = <optimized out>

        protos_no = <optimized out>

        options = 0x664908 "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"

        ret = -1

        seed = 4061548656

        rfd = <optimized out>

        __FUNCTION__ = "main"

thanks
Andy
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From johan at democon.be  Wed Jan 13 16:45:25 2021
From: johan at democon.be (Johan De Clercq)
Date: Wed, 13 Jan 2021 16:45:25 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
In-Reply-To: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
References: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
Message-ID: <PR1PR06MB4683A7F9F61B57F4264632A0A7A90@PR1PR06MB4683.eurprd06.prod.outlook.com>

Firewall is not sip aware, rtprelay via box in dmz

Outlook voor iOS<https://aka.ms/o0ukef> downloaden
________________________________
Van: Users <users-bounces at lists.opensips.org> namens Mark Allen <mark at allenclan.co.uk>
Verzonden: Wednesday, January 13, 2021 5:08:27 PM
Aan: OpenSIPS users mailling list <users at lists.opensips.org>
Onderwerp: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues

Hi all - I've been banging my head against this but not succeeding.

Our setup...

UAC           192.168.x.x
  |
Router        5.x.x.x
  |
(internet)
  |
Firewall      46.x.x.x maps
  |           directly to
OpenSIPS      192.168.x.x      Mid-registrar
  |
Asterisk      192.168.x.x


Current situation:
- UAC can register on Asterisk via OpenSIPS
- UAC can call destination registered on Asterisk on local n/w to Asterisk box
- Destination extension rings and can pick up call
- There is no audio either way & call drops after about 30 secs (Asterisk kills call with "Requested channel not available" because not RTP traffic is reaching destination)

I have tried passing audio through Mediaproxy on OpenSIPS box but with no success. Using Wireshark I can see RTP traffic initiated at both ends, but it doesn't reach the other end either way.

Is there some definitive guide to setting this up correctly or are there specific steps that I need to follow?

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From adierlam at ptgi-ics.com  Wed Jan 13 16:52:20 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Wed, 13 Jan 2021 11:52:20 -0500
Subject: [OpenSIPS-Users] sangoma issue ?
In-Reply-To: <PR1PR06MB46837F7B3E7053F867CFAF49A7A90@PR1PR06MB4683.eurprd06.prod.outlook.com>
References: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>
 <CAKe=zzhgq2vSeqgJeRtzjtn9yRPvf2C+Xd=ez3KoQbhWchaQpQ@mail.gmail.com>
 <PR1PR06MB46837F7B3E7053F867CFAF49A7A90@PR1PR06MB4683.eurprd06.prod.outlook.com>
Message-ID: <CAKe=zzgkPaixKs=wz5VECUTVUm0fhuuhPtBXp4VQDuS=pOK-rg@mail.gmail.com>

Hi John,

I've tested rtpengine's transcoding.   I like the load balancing features.
Though wish it could automatically transcode ( with out flags per call ? )
  And I would think under heavy load, hardware transcoding might scale
better ?  Though maybe dedicated rtpengine servers(pooled) would serve just
as well ?

thanks
Andy


On Wed, Jan 13, 2021 at 11:45 AM Johan De Clercq <johan at democon.be> wrote:

> Why not use rtpengine to do transcoding?
>
> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
> ------------------------------
> *Van:* Users <users-bounces at lists.opensips.org> namens Andy Dierlam <
> adierlam at ptgi-ics.com>
> *Verzonden:* Wednesday, January 13, 2021 4:26:03 PM
> *Aan:* users at lists.opensips.org <users at lists.opensips.org>
> *Onderwerp:* Re: [OpenSIPS-Users] sangoma issue ?
>
> forgot to mention this from:
>
> opensips -V
> version: opensips 3.0.4 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> Q_MALLOC, F_MALLOC, HP_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, sigio_rt, select.
> git revision: d19b20e
> main.c compiled on 14:39:30 Dec 21 2020 with gcc 4.8.5
>
> FYI get the same results from 2.4, 3.0, and 3.1
>
>
>
>
>
> On Tue, Jan 12, 2021 at 3:14 PM Andy Dierlam <adierlam at ptgi-ics.com>
> wrote:
>
> Hello All,
>
> Wondering if anyone can offer some guidance on an issue seemingly with a
> newly setup Sangoma D-500 Transcoding card.
>
> Issue:
> When using transcoding, soon after 1 transcoded call completes, utimer
> task messages until opensips is restarted
> seems like it happens when fetch_dlg_value: looking for <SngTc>
>
> Jan 12 15:05:26 [21578] DBG:tm:timer_routine: timer
> routine:2,tl=0x7f9ce5fd3cc0 next=(nil), timeout=16
>
> Jan 12 15:05:26 [21578] DBG:tm:wait_handler: removing 0x7f9ce5fd3c40 from
> table
>
> Jan 12 15:05:26 [21578] DBG:tm:delete_cell: delete transaction
> 0x7f9ce5fd3c40
>
> Jan 12 15:05:26 [21578] DBG:tm:run_trans_callbacks: trans=0x7f9ce5fd3c40,
> callback type 4096, id 3 entered
>
> Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: destroying dialog
> 0x7f9ce5fcdc50
>
> Jan 12 15:05:26 [21578] DBG:dialog:destroy_dlg: dlg expired or not in list
> - dlg 0x7f9ce5fcdc50 [57:1450509359] with clid
> '3b72b48a67a1eec840e2e1db12f8b37f at x.x.x.x' and tags 'as6d2cf831'
> 'Yjg9IHv3tknz-KgDTUF.K-lleno6X-hI'
>
> Jan 12 15:05:26 [21578] DBG:dialog:run_dlg_callbacks:
> dialog=0x7f9ce5fcdc50, type=2048
>
> Jan 12 15:05:26 [21578] DBG:dialog:fetch_dlg_value: looking for <SngTc>
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 100 ms ago (now 16950 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 200 ms ago (now 17050 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 300 ms ago (now 17150 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 400 ms ago (now 17250 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 490 ms ago (now 17340 ms), delaying execution
>
> Jan 12 15:05:26 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 590 ms ago (now 17440 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 690 ms ago (now 17540 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 790 ms ago (now 17640 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:utimer_ticker: utimer task
> <tm-utimer> already scheduled 890 ms ago (now 17740 ms), delaying execution
>
> Jan 12 15:05:27 [21561] WARNING:core:timer_ticker: timer task <tm-timer>
> already scheduled 990 ms ago (now 17740 ms), delayin
>
>
>
> opensips-cli trap output ( not sure what am looking at here, in case helps
> )
>
> #1  0x00007f9cde9aee5f in sangoma_worker_loop (proc_no=<optimized out>)
> at sngtc_proc.c:46
>
>         req = {type = REQ_FREE_SESSION, response_fd = 78, sng_req =
> {usr_priv = 0x7f9ce8c8f86b <asctime_r+139>, tag = 15, rtcp_enable = 5
> '\005', a = {codec_
>
> id = 32668, ms = 20, host_ip = 0, host_netmask = 2021, host_udp_port = 0},
> b = {codec_id = 9217376, ms = 0, host_ip = 3905485451, host_netmask =
> 32668, host_
>
> udp_port = 20}}, sng_reply = 0x7f9ce5fd2c78}
>
>         rc = 0
>
>         __FUNCTION__ = "sangoma_worker_loop"
>
> #2  0x00000000005045c8 in start_module_procs () at sr_module.c:858
>
>         m = 0x7f9ce7beafa8
>
>         n = 0
>
>         l = 0
>
>         x = <optimized out>
>
>         __FUNCTION__ = "start_module_procs"
>
> #3  0x000000000041e3df in main_loop () at main.c:779
>
>         startup_done = 0x0
>
>         chd_rank = 0
>
>         last_check = 0
>
>         rc = <optimized out>
>
> #4  main (argc=<optimized out>, argv=<optimized out>) at main.c:1479
>
>         c = <optimized out>
>
>         r = 0
>
>         tmp = 0x1 <Address 0x1 out of bounds>
>
>         tmp_len = <optimized out>
>
>         port = <optimized out>
>
>         proto = <optimized out>
>
>         protos_no = <optimized out>
>
>         options = 0x664908
> "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:"
>
>         ret = -1
>
>         seed = 4061548656
>
>         rfd = <optimized out>
>
>         __FUNCTION__ = "main"
>
> thanks
> Andy
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From ag at ag-projects.com  Wed Jan 13 17:29:42 2021
From: ag at ag-projects.com (Adrian Georgescu)
Date: Wed, 13 Jan 2021 14:29:42 -0300
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
In-Reply-To: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
References: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
Message-ID: <56C7C6F6-9789-44C5-A033-112ACB7D48F7@ag-projects.com>

Google search for SIP ALG problem to see if this is relevant for your case.

Regards,
Adrian


> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
> 
> Hi all - I've been banging my head against this but not succeeding.
> 
> Our setup...
> 
> UAC           192.168.x.x
>   | 
> Router        5.x.x.x
>   |
> (internet)
>   | 
> Firewall      46.x.x.x maps
>   |           directly to
> OpenSIPS      192.168.x.x      Mid-registrar
>   |
> Asterisk      192.168.x.x
> 
> 
> Current situation: 
> - UAC can register on Asterisk via OpenSIPS
> - UAC can call destination registered on Asterisk on local n/w to Asterisk box
> - Destination extension rings and can pick up call
> - There is no audio either way & call drops after about 30 secs (Asterisk kills call with "Requested channel not available" because not RTP traffic is reaching destination)
> 
> I have tried passing audio through Mediaproxy on OpenSIPS box but with no success. Using Wireshark I can see RTP traffic initiated at both ends, but it doesn't reach the other end either way.
> 
> Is there some definitive guide to setting this up correctly or are there specific steps that I need to follow? 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From masked at vale.ski  Thu Jan 14 01:55:01 2021
From: masked at vale.ski (bobsy)
Date: Thu, 14 Jan 2021 12:55:01 +1100
Subject: [OpenSIPS-Users] OpenSIPS-CP errors
Message-ID: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>

Hi,

I’m using opensips 3.1.1 from deb repo
And opensips-cp 8.3.1
With nginx and php7.3-fpm
Postgresql for db

And I’ve configured mi_http for json…

I’m receiving a few errors like this:

Failed to issue total count query, error message : Array ( [0] => 42P01 [1] => 7 [2] => ERROR: relation "cc_agents" does not exist LINE 2: from cc_agents ^ ) [select count(id) from cc_agents]

On the call centre page and on the dialog page


2021/01/14 12:50:29 [error] 3849#3849: *289 FastCGI sent in stderr: "PHP message: PHP Warning:  Creating default object from empty value in /var/www/opensips-cp/config/tools/system/dialog/local.inc.php on line 27PHP message: PHP Notice:  Array to string conversion in /var/www/opensips-cp/web/tools/system/dialog/dialog.php on line 112" while reading response header from upstream, client: 119.18.18.117, server: opensips.vale.ski, request: "GET /tools/system/dialog/dialog.php HTTP/1.1", upstream: "fastcgi://unix:/run/php/php7.3-fpm.sock:", host: "opensips.vale.ski", referrer: "http://opensips.vale.ski/menu.php”

I get this error on that page:

MI command failed with code -32601 (Method not found)

Any insight would be greatly appreciated.


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From masked at vale.ski  Thu Jan 14 01:58:59 2021
From: masked at vale.ski (bobsy)
Date: Thu, 14 Jan 2021 12:58:59 +1100
Subject: [OpenSIPS-Users] OpenSIPS-CP errors
In-Reply-To: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>
References: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>
Message-ID: <E75C508F-8427-4045-85F7-A42EBF21B3E7@vale.ski>

I also get this in OpenSIPS

 DBG:httpd:answer_to_connection: START *** cls=(nil), connection=0x55d766a1b220, url=/mi, method=POST, versio=HTTP/1.1, upload_data[48]=0x55d766a229cc, *con_cls=0x7f53f1f2d4d0
Jan 14 12:57:26 [27593] DBG:httpd:answer_to_connection: NOT a regular POST :o)
Jan 14 12:57:26 [27593] DBG:httpd:getConnectionHeader: key=[Host] value=[127.0.0.1:8888]
Jan 14 12:57:26 [27593] DBG:httpd:getConnectionHeader: Accept=*/*
Jan 14 12:57:26 [27593] DBG:httpd:getConnectionHeader: Content-Type=application/json
Jan 14 12:57:26 [27593] DBG:httpd:getConnectionHeader: Content-Length=48
Jan 14 12:57:26 [27593] DBG:httpd:answer_to_connection: got ContentType [3] with len [48]: {"jsonrpc":"2.0","id":1,"method":"dr_gw_status"}\nJan 14 12:57:26 [27593] DBG:httpd:answer_to_connection: START *** cls=(nil), connection=0x55d766a1b220, url=/mi, method=POST, versio=HTTP/1.1, upload_data[0]=(nil), *con_cls=0x7f53f1f2d4d0
Jan 14 12:57:26 [27593] DBG:httpd:answer_to_connection: normalised_url=[]
Jan 14 12:57:26 [27593] DBG:mi_http:mi_json_answer_to_connection: START *** cls=(nil), connection=0x55d766a1b220, url=, method=POST, version=HTTP/1.1, upload_data[48]=0x7f53f1f2b299, *con_cls=0x7f53f1f2d468
Jan 14 12:57:26 [27593] DBG:mi_http:mi_http_run_mi_cmd: got command=dr_gw_status
Jan 14 12:57:26 [27593] ERROR:core:handle_mi_request: Command not found

> On 14 Jan 2021, at 12:55 pm, bobsy <masked at vale.ski> wrote:
> 
> Hi,
> 
> I’m using opensips 3.1.1 from deb repo
> And opensips-cp 8.3.1
> With nginx and php7.3-fpm
> Postgresql for db
> 
> And I’ve configured mi_http for json…
> 
> I’m receiving a few errors like this:
> 
> Failed to issue total count query, error message : Array ( [0] => 42P01 [1] => 7 [2] => ERROR: relation "cc_agents" does not exist LINE 2: from cc_agents ^ ) [select count(id) from cc_agents]
> 
> On the call centre page and on the dialog page
> 
> 
> 2021/01/14 12:50:29 [error] 3849#3849: *289 FastCGI sent in stderr: "PHP message: PHP Warning:  Creating default object from empty value in /var/www/opensips-cp/config/tools/system/dialog/local.inc.php on line 27PHP message: PHP Notice:  Array to string conversion in /var/www/opensips-cp/web/tools/system/dialog/dialog.php on line 112" while reading response header from upstream, client: 119.18.18.117, server: opensips.vale.ski, request: "GET /tools/system/dialog/dialog.php HTTP/1.1", upstream: "fastcgi://unix:/run/php/php7.3-fpm.sock: <fastcgi://unix/run/php/php7.3-fpm.sock:>", host: "opensips.vale.ski", referrer: "http://opensips.vale.ski/menu.php <http://opensips.vale.ski/menu.php>”
> 
> I get this error on that page:
> 
> MI command failed with code -32601 (Method not found)
> 
> Any insight would be greatly appreciated.
> 
> 

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From masked at vale.ski  Thu Jan 14 02:07:05 2021
From: masked at vale.ski (bobsy)
Date: Thu, 14 Jan 2021 13:07:05 +1100
Subject: [OpenSIPS-Users] OpenSIPS-CP errors
In-Reply-To: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>
References: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>
Message-ID: <AF02A27A-AA5E-43CD-B15A-D562F4E6030E@vale.ski>

While I’m at it when I try to create a script using osipsconfig I get this :


          Config generated : /etc/opensips/opensips_loadbalancer_2021-1-14_13:6:
26.cfg = FAILED (). Press any key to continue


> On 14 Jan 2021, at 12:55 pm, bobsy via Users <users at lists.opensips.org> wrote:
> 
> Hi,
> 
> I’m using opensips 3.1.1 from deb repo
> And opensips-cp 8.3.1
> With nginx and php7.3-fpm
> Postgresql for db
> 
> And I’ve configured mi_http for json…
> 
> I’m receiving a few errors like this:
> 
> Failed to issue total count query, error message : Array ( [0] => 42P01 [1] => 7 [2] => ERROR: relation "cc_agents" does not exist LINE 2: from cc_agents ^ ) [select count(id) from cc_agents]
> 
> On the call centre page and on the dialog page
> 
> 
> 2021/01/14 12:50:29 [error] 3849#3849: *289 FastCGI sent in stderr: "PHP message: PHP Warning:  Creating default object from empty value in /var/www/opensips-cp/config/tools/system/dialog/local.inc.php on line 27PHP message: PHP Notice:  Array to string conversion in /var/www/opensips-cp/web/tools/system/dialog/dialog.php on line 112" while reading response header from upstream, client: 119.18.18.117, server: opensips.vale.ski, request: "GET /tools/system/dialog/dialog.php HTTP/1.1", upstream: "fastcgi://unix:/run/php/php7.3-fpm.sock: <fastcgi://unix/run/php/php7.3-fpm.sock:>", host: "opensips.vale.ski", referrer: "http://opensips.vale.ski/menu.php <http://opensips.vale.ski/menu.php>”
> 
> I get this error on that page:
> 
> MI command failed with code -32601 (Method not found)
> 
> Any insight would be greatly appreciated.
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From masked at vale.ski  Thu Jan 14 02:12:18 2021
From: masked at vale.ski (bobsy)
Date: Thu, 14 Jan 2021 13:12:18 +1100
Subject: [OpenSIPS-Users] OpenSIPS-CP errors
In-Reply-To: <AF02A27A-AA5E-43CD-B15A-D562F4E6030E@vale.ski>
References: <E1A0CD78-9B6F-4383-A126-25702BB3E757@vale.ski>
 <AF02A27A-AA5E-43CD-B15A-D562F4E6030E@vale.ski>
Message-ID: <6D9981D9-82ED-49F6-8E50-5650F4DE99DA@vale.ski>

Okay m4 dependency was missing.

> On 14 Jan 2021, at 1:07 pm, bobsy <masked at vale.ski> wrote:
> 
> While I’m at it when I try to create a script using osipsconfig I get this :
> 
> 
>           Config generated : /etc/opensips/opensips_loadbalancer_2021-1-14_13:6:
> 26.cfg = FAILED (). Press any key to continue
> 
> 
>> On 14 Jan 2021, at 12:55 pm, bobsy via Users <users at lists.opensips.org <mailto:users at lists.opensips.org>> wrote:
>> 
>> Hi,
>> 
>> I’m using opensips 3.1.1 from deb repo
>> And opensips-cp 8.3.1
>> With nginx and php7.3-fpm
>> Postgresql for db
>> 
>> And I’ve configured mi_http for json…
>> 
>> I’m receiving a few errors like this:
>> 
>> Failed to issue total count query, error message : Array ( [0] => 42P01 [1] => 7 [2] => ERROR: relation "cc_agents" does not exist LINE 2: from cc_agents ^ ) [select count(id) from cc_agents]
>> 
>> On the call centre page and on the dialog page
>> 
>> 
>> 2021/01/14 12:50:29 [error] 3849#3849: *289 FastCGI sent in stderr: "PHP message: PHP Warning:  Creating default object from empty value in /var/www/opensips-cp/config/tools/system/dialog/local.inc.php on line 27PHP message: PHP Notice:  Array to string conversion in /var/www/opensips-cp/web/tools/system/dialog/dialog.php on line 112" while reading response header from upstream, client: 119.18.18.117, server: opensips.vale.ski, request: "GET /tools/system/dialog/dialog.php HTTP/1.1", upstream: "fastcgi://unix:/run/php/php7.3-fpm.sock: <fastcgi://unix/run/php/php7.3-fpm.sock:>", host: "opensips.vale.ski", referrer: "http://opensips.vale.ski/menu.php <http://opensips.vale.ski/menu.php>”
>> 
>> I get this error on that page:
>> 
>> MI command failed with code -32601 (Method not found)
>> 
>> Any insight would be greatly appreciated.
>> 
>> 
>> _______________________________________________
>> 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 mark at allenclan.co.uk  Thu Jan 14 08:22:16 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 14 Jan 2021 08:22:16 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
In-Reply-To: <56C7C6F6-9789-44C5-A033-112ACB7D48F7@ag-projects.com>
References: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
 <56C7C6F6-9789-44C5-A033-112ACB7D48F7@ag-projects.com>
Message-ID: <CADaqb1s70L21dEfgp8WYmbjvTOnL33FUwOu5NomYffQZn2veuQ@mail.gmail.com>

Thanks Adrian

The firewall has SIP-ALG disabled and just forwards ports from externally
to where they need to be internally - so ports 5060 and 10000 - 65535 of
46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)

On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu <ag at ag-projects.com> wrote:

> Google search for SIP ALG problem to see if this is relevant for your case.
>
> Regards,
> Adrian
>
>
> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
>
> Hi all - I've been banging my head against this but not succeeding.
>
> Our setup...
>
> UAC           192.168.x.x
>   |
> Router        5.x.x.x
>   |
> (internet)
>   |
> Firewall      46.x.x.x maps
>   |           directly to
> OpenSIPS      192.168.x.x      Mid-registrar
>   |
> Asterisk      192.168.x.x
>
>
> Current situation:
> - UAC can register on Asterisk via OpenSIPS
> - UAC can call destination registered on Asterisk on local n/w to Asterisk
> box
> - Destination extension rings and can pick up call
> - There is no audio either way & call drops after about 30 secs (Asterisk
> kills call with "Requested channel not available" because not RTP traffic
> is reaching destination)
>
> I have tried passing audio through Mediaproxy on OpenSIPS box but with no
> success. Using Wireshark I can see RTP traffic initiated at both ends, but
> it doesn't reach the other end either way.
>
> Is there some definitive guide to setting this up correctly or are there
> specific steps that I need to follow?
>
> _______________________________________________
> 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 david.villasmil.work at gmail.com  Thu Jan 14 11:57:38 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 14 Jan 2021 11:57:38 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
In-Reply-To: <CADaqb1s70L21dEfgp8WYmbjvTOnL33FUwOu5NomYffQZn2veuQ@mail.gmail.com>
References: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
 <56C7C6F6-9789-44C5-A033-112ACB7D48F7@ag-projects.com>
 <CADaqb1s70L21dEfgp8WYmbjvTOnL33FUwOu5NomYffQZn2veuQ@mail.gmail.com>
Message-ID: <CAFGRPVoeNLwdJOEHvyAMHYn02wE1c2hEcAktoHaegHzQg0u+gA@mail.gmail.com>

Check out what IPs are offered in the SDPs in asterisk. Make sure they’re
both public IPs.
If you only have 1 asterisk, forwarding the rtp port range configured in
asterisk from the firewall to asterisk should do it.


On Thu, 14 Jan 2021 at 08:23, Mark Allen <mark at allenclan.co.uk> wrote:

> Thanks Adrian
>
> The firewall has SIP-ALG disabled and just forwards ports from externally
> to where they need to be internally - so ports 5060 and 10000 - 65535 of
> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)
>
> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu <ag at ag-projects.com> wrote:
>
>> Google search for SIP ALG problem to see if this is relevant for your
>> case.
>>
>> Regards,
>> Adrian
>>
>>
>> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
>>
>> Hi all - I've been banging my head against this but not succeeding.
>>
>> Our setup...
>>
>> UAC           192.168.x.x
>>   |
>> Router        5.x.x.x
>>   |
>> (internet)
>>   |
>> Firewall      46.x.x.x maps
>>   |           directly to
>> OpenSIPS      192.168.x.x      Mid-registrar
>>   |
>> Asterisk      192.168.x.x
>>
>>
>> Current situation:
>> - UAC can register on Asterisk via OpenSIPS
>> - UAC can call destination registered on Asterisk on local n/w to
>> Asterisk box
>> - Destination extension rings and can pick up call
>> - There is no audio either way & call drops after about 30 secs (Asterisk
>> kills call with "Requested channel not available" because not RTP
>> traffic is reaching destination)
>>
>> I have tried passing audio through Mediaproxy on OpenSIPS box but with no
>> success. Using Wireshark I can see RTP traffic initiated at both ends, but
>> it doesn't reach the other end either way.
>>
>> Is there some definitive guide to setting this up correctly or are there
>> specific steps that I need to follow?
>>
>> _______________________________________________
>> 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
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From mark at allenclan.co.uk  Thu Jan 14 14:03:12 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Thu, 14 Jan 2021 14:03:12 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1 & NAT issues
In-Reply-To: <CAFGRPVoeNLwdJOEHvyAMHYn02wE1c2hEcAktoHaegHzQg0u+gA@mail.gmail.com>
References: <CADaqb1tww9BySVsNNkuJpzLEAtJYZePhJU4gNVBaiUrCvkAPOw@mail.gmail.com>
 <56C7C6F6-9789-44C5-A033-112ACB7D48F7@ag-projects.com>
 <CADaqb1s70L21dEfgp8WYmbjvTOnL33FUwOu5NomYffQZn2veuQ@mail.gmail.com>
 <CAFGRPVoeNLwdJOEHvyAMHYn02wE1c2hEcAktoHaegHzQg0u+gA@mail.gmail.com>
Message-ID: <CADaqb1s1FYexCP15yfgweBLB8U_EgjFnnPuWJ+Y1nx6Q+Cw7SA@mail.gmail.com>

Thanks for the responses. They helped me exclude some things. I've managed
to make progress and pinned down the lack of audio to a misconfiguration of
Mediaproxy. Two-way audio through double-nat / firewall is working but goes
silent after about 60 seconds connected and Asterisk kills the connection
31 seconds later due to lack of RTP activity for the last 31 seconds

On Thu, 14 Jan 2021 at 12:00, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Check out what IPs are offered in the SDPs in asterisk. Make sure they’re
> both public IPs.
> If you only have 1 asterisk, forwarding the rtp port range configured in
> asterisk from the firewall to asterisk should do it.
>
>
> On Thu, 14 Jan 2021 at 08:23, Mark Allen <mark at allenclan.co.uk> wrote:
>
>> Thanks Adrian
>>
>> The firewall has SIP-ALG disabled and just forwards ports from externally
>> to where they need to be internally - so ports 5060 and 10000 - 65535 of
>> 46.x.x.x are mapped to 192.168.x.x (the OpenSIPS box)
>>
>> On Wed, 13 Jan 2021 at 17:32, Adrian Georgescu <ag at ag-projects.com>
>> wrote:
>>
>>> Google search for SIP ALG problem to see if this is relevant for your
>>> case.
>>>
>>> Regards,
>>> Adrian
>>>
>>>
>>> On 13 Jan 2021, at 13:08, Mark Allen <mark at allenclan.co.uk> wrote:
>>>
>>> Hi all - I've been banging my head against this but not succeeding.
>>>
>>> Our setup...
>>>
>>> UAC           192.168.x.x
>>>   |
>>> Router        5.x.x.x
>>>   |
>>> (internet)
>>>   |
>>> Firewall      46.x.x.x maps
>>>   |           directly to
>>> OpenSIPS      192.168.x.x      Mid-registrar
>>>   |
>>> Asterisk      192.168.x.x
>>>
>>>
>>> Current situation:
>>> - UAC can register on Asterisk via OpenSIPS
>>> - UAC can call destination registered on Asterisk on local n/w to
>>> Asterisk box
>>> - Destination extension rings and can pick up call
>>> - There is no audio either way & call drops after about 30 secs
>>> (Asterisk kills call with "Requested channel not available" because not
>>> RTP traffic is reaching destination)
>>>
>>> I have tried passing audio through Mediaproxy on OpenSIPS box but with
>>> no success. Using Wireshark I can see RTP traffic initiated at both ends,
>>> but it doesn't reach the other end either way.
>>>
>>> Is there some definitive guide to setting this up correctly or are there
>>> specific steps that I need to follow?
>>>
>>> _______________________________________________
>>> 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
>>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From Sunil.More at samespace.com  Fri Jan 15 07:52:01 2021
From: Sunil.More at samespace.com (Sunil More)
Date: Fri, 15 Jan 2021 13:22:01 +0530
Subject: [OpenSIPS-Users] xlog message sent to homer 7 via HEP has type
	as ERROR
In-Reply-To: <81085add-60f5-b5d4-209b-314e7930cb71@opensips.org>
References: <CAAP4-XkNOCEVn-DeGGAef7_QLFB7GZ0ggacQ9BaAeRrADTWtpA@mail.gmail.com>
 <81085add-60f5-b5d4-209b-314e7930cb71@opensips.org>
Message-ID: <CAAP4-XnnV7rKTnmkDPiz5G9AR9EPSs3Ub0HHLR+_6wqv-CJ60Q@mail.gmail.com>

Hello Liviu Chircu and Team,

I have recompiled and checked with said commit of opensips
opensips -V
version: opensips 3.1.1 (x86_64/linux)
git revision: 2212865f1
main.c compiled on 07:18:08 Jan 15 2021 with gcc 7

Now the logs are seen as NOTICE and not ERROR.

L_INFO does not make it to HEP logs
L_ALERT makes it to HEP as NOTICE
and other logs used as xlog("Anything ") also make it as NOTICE.

should they look like this or L_INFO should be seen as INFO while L_ALERT
seen as ALERT .

Sunil More

Manager - DevOps

91 95033 38275

sunil.more at samespace.com



<https://www.samespace.com/>


On Wed, Jan 13, 2021 at 7:55 PM Liviu Chircu <liviu at opensips.org> wrote:

> On 13.01.2021 15:24, Sunil More wrote:
> > I am using opensips 3.1.1 along with homer 7 and I could observe that
> > logs are going to Homer with type as ERROR.
> >
> > Here's opensips -V
> > version: opensips 3.1.1 (x86_64/linux)
> > git revision: 229ec0793
> > main.c compiled on 10:46:42 Jan  7 2021 with gcc 7
> >
> Hi,
>
> This has been fixed on Dec 10th [1], so you have 3 options:
>
> * pull latest 3.1 source code and rebuild OpenSIPS
> * install nightly 3.1 packages
> * wait until 3.1.2 release
>
> [1]: https://github.com/OpenSIPS/opensips/commit/2212865f19d
>
> Cheers,
>
> --
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 
 <http://www.samespace.com>



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From ag at ag-projects.com  Fri Jan 15 17:07:34 2021
From: ag at ag-projects.com (Adrian Georgescu)
Date: Fri, 15 Jan 2021 14:07:34 -0300
Subject: [OpenSIPS-Users] OpenSIPS as simple SIP proxy
In-Reply-To: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>
References: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>
Message-ID: <0EB460B0-0E5D-4B41-A730-6C99A41BF5F6@ag-projects.com>


You can install a pre-configured SIP Proxy as a start point, like OpenSIPS.

Installation instructions are available.

here <http://download.ag-projects.com/OpenSIPS/INSTALL>

OpenSIPS configuration debian package is available here <https://docs-new.sipthor.net/w/debian_package_repositories/>.

The package name is opensips-config-light

Regards,
Adrian


> On 13 Jan 2021, at 04:16, Conor Power <conorjpower at hotmail.com> wrote:
> 
> Hi,
>    Apologies for the noob question but I'm hoping someone can point me in the right direction.
> 
> I am trying to use OpenSIPS as a simple proxy to proxy all calls to another SIP endpoint and back again to the original client. The only role of the OpenSIPS server is to function as the proxy and it is for all requests.
> 
> I have OpenSIPS up and running and can see the requests coming inbound using ngrep but I've had no success proxying the requests.
> 
> I added a sethostport() call in the config file but really am not sure where or how it fits in.
> 
> If someone might point me to a simple config file that would be used for such a proxy setup, it would be greatly appreciated.
> 
> Conor
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
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From gmaruzz at gmail.com  Sat Jan 16 19:13:35 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 16 Jan 2021 20:13:35 +0100
Subject: [OpenSIPS-Users] [SOLVED] 3.1 How to build DEB Debian Packages from
	source (Stretch)
Message-ID: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>

Hello my friends,

I know I'm slow, it took me much time to find the way to build deb packages
from source.

So, without further ado, my receipt (on Stretch, may works on Buster too)
-BOTH git clone AND wget are required, it is not a typo- :

=========

mkdir /usr/src/OPENSIPS
cd /usr/src/OPENSIPS

git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
wget https://github.com/OpenSIPS/opensips/archive/3.1.tar.gz -O
opensips_3.1.1.orig.tar.gz

apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
libmongoc-dev

cd opensips_3_1

make deb
cd ..

ls

dpkg -i opensips_3.1.1-1_amd64.deb
opensips-dialplan-module_3.1.1-1_amd64.deb
opensips-http-modules_3.1.1-1_amd64.deb
opensips-lua-module_3.1.1-1_amd64.deb
opensips-memcached-module_3.1.1-1_amd64.deb
opensips-postgres-module_3.1.1-1_amd64.deb
opensips-presence-modules_3.1.1-1_amd64.deb
opensips-redis-module_3.1.1-1_amd64.deb
opensips-regex-module_3.1.1-1_amd64.deb
opensips-restclient-module_3.1.1-1_amd64.deb
opensips-sqlite-module_3.1.1-1_amd64.deb
opensips-tlsmgm-module_3.1.1-1_amd64.deb
opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
opensips-xml-module_3.1.1-1_amd64.deb
opensips-xmlrpc-module_3.1.1-1_amd64.deb

=========

Obviously, you can choose which deb packages to install :)

HTH

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From osas at voipembedded.com  Sat Jan 16 19:41:38 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Sat, 16 Jan 2021 14:41:38 -0500
Subject: [OpenSIPS-Users] [SOLVED] 3.1 How to build DEB Debian Packages
 from source (Stretch)
In-Reply-To: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>
References: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>
Message-ID: <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>

Hello Giovanni,

After you clone the repo, you need to build the tar from it:
make deb-orig-tar
You can skip the wget.

If you modify the source, rebuild the tar and then rebuild the deb package.

Also, when building the package, use the following options (so you
won't have troubles with signed packages):
make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"

-ovidiu

On Sat, Jan 16, 2021 at 2:15 PM Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>
> Hello my friends,
>
> I know I'm slow, it took me much time to find the way to build deb packages from source.
>
> So, without further ado, my receipt (on Stretch, may works on Buster too)
> -BOTH git clone AND wget are required, it is not a typo- :
>
> =========
>
> mkdir /usr/src/OPENSIPS
> cd /usr/src/OPENSIPS
>
> git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
> wget https://github.com/OpenSIPS/opensips/archive/3.1.tar.gz -O opensips_3.1.1.orig.tar.gz
>
> apt-get install bison debhelper default-libmysqlclient-dev dh-systemd dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev libmongoc-dev
>
> cd opensips_3_1
>
> make deb
> cd ..
>
> ls
>
> dpkg -i opensips_3.1.1-1_amd64.deb opensips-dialplan-module_3.1.1-1_amd64.deb opensips-http-modules_3.1.1-1_amd64.deb opensips-lua-module_3.1.1-1_amd64.deb opensips-memcached-module_3.1.1-1_amd64.deb opensips-postgres-module_3.1.1-1_amd64.deb opensips-presence-modules_3.1.1-1_amd64.deb opensips-redis-module_3.1.1-1_amd64.deb opensips-regex-module_3.1.1-1_amd64.deb opensips-restclient-module_3.1.1-1_amd64.deb opensips-sqlite-module_3.1.1-1_amd64.deb opensips-tlsmgm-module_3.1.1-1_amd64.deb opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb opensips-xml-module_3.1.1-1_amd64.deb opensips-xmlrpc-module_3.1.1-1_amd64.deb
>
> =========
>
> Obviously, you can choose which deb packages to install :)
>
> HTH
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com


From gmaruzz at gmail.com  Sat Jan 16 19:46:07 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 16 Jan 2021 20:46:07 +0100
Subject: [OpenSIPS-Users] [SOLVED] 3.1 How to build DEB Debian Packages
 from source (Stretch)
In-Reply-To: <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>
References: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>
 <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>
Message-ID: <CALXCt0p7ch78aZh6W5vf+Uixs=Qp8FeXrCLEWkCftXBn=ZD1iQ@mail.gmail.com>

Ciao Ovidiu!!!!

So many thanks, sure your suggestion seems the right way!


answered from mobile, please pardon terseness and typos,
-giovanni

On Sat, Jan 16, 2021, 20:41 Ovidiu Sas <osas at voipembedded.com> wrote:

> Hello Giovanni,
>
> After you clone the repo, you need to build the tar from it:
> make deb-orig-tar
> You can skip the wget.
>
> If you modify the source, rebuild the tar and then rebuild the deb package.
>
> Also, when building the package, use the following options (so you
> won't have troubles with signed packages):
> make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"
>
> -ovidiu
>
> On Sat, Jan 16, 2021 at 2:15 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> >
> > Hello my friends,
> >
> > I know I'm slow, it took me much time to find the way to build deb
> packages from source.
> >
> > So, without further ado, my receipt (on Stretch, may works on Buster too)
> > -BOTH git clone AND wget are required, it is not a typo- :
> >
> > =========
> >
> > mkdir /usr/src/OPENSIPS
> > cd /usr/src/OPENSIPS
> >
> > git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
> > wget https://github.com/OpenSIPS/opensips/archive/3.1.tar.gz -O
> opensips_3.1.1.orig.tar.gz
> >
> > apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
> dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
> libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
> libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
> libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
> libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
> pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
> libmongoc-dev
> >
> > cd opensips_3_1
> >
> > make deb
> > cd ..
> >
> > ls
> >
> > dpkg -i opensips_3.1.1-1_amd64.deb
> opensips-dialplan-module_3.1.1-1_amd64.deb
> opensips-http-modules_3.1.1-1_amd64.deb
> opensips-lua-module_3.1.1-1_amd64.deb
> opensips-memcached-module_3.1.1-1_amd64.deb
> opensips-postgres-module_3.1.1-1_amd64.deb
> opensips-presence-modules_3.1.1-1_amd64.deb
> opensips-redis-module_3.1.1-1_amd64.deb
> opensips-regex-module_3.1.1-1_amd64.deb
> opensips-restclient-module_3.1.1-1_amd64.deb
> opensips-sqlite-module_3.1.1-1_amd64.deb
> opensips-tlsmgm-module_3.1.1-1_amd64.deb
> opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
> opensips-xml-module_3.1.1-1_amd64.deb
> opensips-xmlrpc-module_3.1.1-1_amd64.deb
> >
> > =========
> >
> > Obviously, you can choose which deb packages to install :)
> >
> > HTH
> >
> > -giovanni
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> > cell: +39 347 266 56 18
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
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From spanda at 3clogic.com  Mon Jan 18 07:52:46 2021
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 18 Jan 2021 13:22:46 +0530
Subject: [OpenSIPS-Users] Need some help while configuring opensips 3.1
	with homer 7 .
In-Reply-To: <CALf-Q466y43TCNteqUJeX0FWeDncwCB08V0Qe9PhV_LK9F4VKA@mail.gmail.com>
References: <CALf-Q46v+Nx5tf8XNTX-Ma9DFnqU_nUSqEx42W-0m_LSikBzmg@mail.gmail.com>
 <CAKN+pHYSjone=u+YS9-ANwPEV5bgNsJn25dJAHWeqwr642xGzg@mail.gmail.com>
 <CALf-Q466y43TCNteqUJeX0FWeDncwCB08V0Qe9PhV_LK9F4VKA@mail.gmail.com>
Message-ID: <CALf-Q47+LfqtQDf09+bZ71-DOTZOwe-NgikbriS5-Q6Yp8vFmg@mail.gmail.com>

Hi ,

Is there any update on this? Please do help .

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


On Wed, Jan 13, 2021 at 11:18 AM Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi Lorenzo ,
>
> Need some understanding about the exported function  correlate in
> proto_hep module .
>
> The documentation says through this we can correlate two calls .
>
> correlate("hep_dst", "correlation-no-1",$var(cor1),"correlation-no-2", $var(cor2));
>
>
> My scenario .
> user1 -- opensips --- Freeswitch --- opensips --- user2
>
> Here freeswitch behaves like a B2B UA . So when user1 calls to user2 there
> is 2 call-id in both leg .
> I want to correlate both the calls in a single flow . Freeswitch shares an
> UUID in both the leg to identify these two calls . But in homer side the
> call-id is he session id .
>
> How I will do this ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Thu, Dec 17, 2020 at 6:19 PM Lorenzo Mangani <lorenzo.mangani at gmail.com>
> wrote:
>
>> Hi Sasmita,
>>
>> Please open an issue on the homer repository and we'll gladly assist you
>> or any OpenSIPS community member :)
>>
>> https://github.com/sipcapture/homer/issues
>>
>>
>> Kind Regards,
>>
>> Lorenzo Mangani
>>
>> QXIP BV - Capture Engineering
>> Amsterdam, The Netherlands
>>
>> CONFIDENTIALITY NOTICE: This e-mail message, including any attachments, is
>> for the sole use of the intended recipient(s) and may contain confidential or
>> legally privileged information. Any unauthorized review, use, disclosure or
>> distribution is prohibited. If you are not the intended recipient, please contact
>> the sender by reply e-mail and destroy all copies of this original message.
>>
>
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From ztzozon at gmail.com  Mon Jan 18 08:41:34 2021
From: ztzozon at gmail.com (zozon)
Date: Mon, 18 Jan 2021 01:41:34 -0700 (MST)
Subject: [OpenSIPS-Users] Rate_Cacher
Message-ID: <1610959294662-0.post@n2.nabble.com>

Hi guys,
I'm currently considering to use cgrates, but the functionality of
rate_cacher would be enough for me. therefore I was wondering, when the
beta/production version could be expected. Any milestones you could share? 



--
Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html


From spanda at 3clogic.com  Mon Jan 18 11:04:31 2021
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 18 Jan 2021 16:34:31 +0530
Subject: [OpenSIPS-Users] Question regarding auth_jwt module .
Message-ID: <CALf-Q46b1K24Q-bw3Qn9OB2zfx+VAnE2ciNEXp7vb1t4E0xt2A@mail.gmail.com>

Hi All ,

First time I am going to use auth_jwt module in opensips 3.1 .
I have gone through the ppt and video of 2020 opensips Summit .

https://www.opensips.org/events/Summit-2020Distributed/assets/presentations/Vlad_Paiu-Opensips%20Jwt%202.pdf

https://www.youtube.com/watch?v=T-_uN7mdffE

I have a confusion , please clear this . When we integrate jwt
authentication in jssip . that will get the jwt token from a http server
(example ) . And when it registers itself with Opensips in Register request
it will send the Authorization header with that verified token .

But my question is , how opensips will get the profile tag and the secret
for the token in its mysql table ? Something I am missing . I can't add the
data in mysql at runtime .
Then how opensips will decode the token and Authenticate the register
request .?

Please do suggest .


*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From gmaruzz at gmail.com  Mon Jan 18 13:06:37 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 18 Jan 2021 14:06:37 +0100
Subject: [OpenSIPS-Users] [SOLVED] 3.1 How to build DEB Debian Packages
 from source (Stretch)
In-Reply-To: <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>
References: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>
 <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>
Message-ID: <CALXCt0oFJB_s8hw_9w63yB54CD01N3iNJzP74b6POiAQANTH2g@mail.gmail.com>

My friends, Ovidiu,

so, see the following two scripts, one for first time building deb
packages, second script for updating such packages.

Can you all please give hints, corrections, etc?

FIRST TIME BUILD DEB PACKAGES:
#=================
apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
libmongoc-dev

mkdir /usr/src/OPENSIPS_DEBS
cd /usr/src/OPENSIPS_DEBS

git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1

cd opensips_3_1

make deb-orig-tar
make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"

cd ..
ls

dpkg -i opensips_3.1.1-1_amd64.deb
opensips-dialplan-module_3.1.1-1_amd64.deb
opensips-http-modules_3.1.1-1_amd64.deb
opensips-lua-module_3.1.1-1_amd64.deb
opensips-memcached-module_3.1.1-1_amd64.deb
opensips-postgres-module_3.1.1-1_amd64.deb
opensips-presence-modules_3.1.1-1_amd64.deb
opensips-redis-module_3.1.1-1_amd64.deb
opensips-regex-module_3.1.1-1_amd64.deb
opensips-restclient-module_3.1.1-1_amd64.deb
opensips-sqlite-module_3.1.1-1_amd64.deb
opensips-tlsmgm-module_3.1.1-1_amd64.deb
opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
opensips-xml-module_3.1.1-1_amd64.deb
opensips-xmlrpc-module_3.1.1-1_amd64.deb

#=================

UPDATE DEB PACKAGES (keeping local source changes):
#=================
cd /usr/src/OPENSIPS_DEBS/opensips_3_1

git stash ; git pull ; git stash apply

make deb-orig-tar
make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"

cd ..
ls

dpkg -i opensips_3.1.1-1_amd64.deb
opensips-dialplan-module_3.1.1-1_amd64.deb
opensips-http-modules_3.1.1-1_amd64.deb
opensips-lua-module_3.1.1-1_amd64.deb
opensips-memcached-module_3.1.1-1_amd64.deb
opensips-postgres-module_3.1.1-1_amd64.deb
opensips-presence-modules_3.1.1-1_amd64.deb
opensips-redis-module_3.1.1-1_amd64.deb
opensips-regex-module_3.1.1-1_amd64.deb
opensips-restclient-module_3.1.1-1_amd64.deb
opensips-sqlite-module_3.1.1-1_amd64.deb
opensips-tlsmgm-module_3.1.1-1_amd64.deb
opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
opensips-xml-module_3.1.1-1_amd64.deb
opensips-xmlrpc-module_3.1.1-1_amd64.deb

#=================


On Sat, Jan 16, 2021 at 8:41 PM Ovidiu Sas <osas at voipembedded.com> wrote:

> Hello Giovanni,
>
> After you clone the repo, you need to build the tar from it:
> make deb-orig-tar
> You can skip the wget.
>
> If you modify the source, rebuild the tar and then rebuild the deb package.
>
> Also, when building the package, use the following options (so you
> won't have troubles with signed packages):
> make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"
>
> -ovidiu
>
> On Sat, Jan 16, 2021 at 2:15 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
> >
> > Hello my friends,
> >
> > I know I'm slow, it took me much time to find the way to build deb
> packages from source.
> >
> > So, without further ado, my receipt (on Stretch, may works on Buster too)
> > -BOTH git clone AND wget are required, it is not a typo- :
> >
> > =========
> >
> > mkdir /usr/src/OPENSIPS
> > cd /usr/src/OPENSIPS
> >
> > git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
> > wget https://github.com/OpenSIPS/opensips/archive/3.1.tar.gz -O
> opensips_3.1.1.orig.tar.gz
> >
> > apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
> dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
> libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
> libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
> libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
> libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
> pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
> libmongoc-dev
> >
> > cd opensips_3_1
> >
> > make deb
> > cd ..
> >
> > ls
> >
> > dpkg -i opensips_3.1.1-1_amd64.deb
> opensips-dialplan-module_3.1.1-1_amd64.deb
> opensips-http-modules_3.1.1-1_amd64.deb
> opensips-lua-module_3.1.1-1_amd64.deb
> opensips-memcached-module_3.1.1-1_amd64.deb
> opensips-postgres-module_3.1.1-1_amd64.deb
> opensips-presence-modules_3.1.1-1_amd64.deb
> opensips-redis-module_3.1.1-1_amd64.deb
> opensips-regex-module_3.1.1-1_amd64.deb
> opensips-restclient-module_3.1.1-1_amd64.deb
> opensips-sqlite-module_3.1.1-1_amd64.deb
> opensips-tlsmgm-module_3.1.1-1_amd64.deb
> opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
> opensips-xml-module_3.1.1-1_amd64.deb
> opensips-xmlrpc-module_3.1.1-1_amd64.deb
> >
> > =========
> >
> > Obviously, you can choose which deb packages to install :)
> >
> > HTH
> >
> > -giovanni
> >
> > --
> > Sincerely,
> >
> > Giovanni Maruzzelli
> > OpenTelecom.IT
> > cell: +39 347 266 56 18
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From mark at allenclan.co.uk  Mon Jan 18 15:57:08 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Mon, 18 Jan 2021 15:57:08 +0000
Subject: [OpenSIPS-Users] OpenSIPS 3.1, mid_registrar and NAT handling
Message-ID: <CADaqb1t6Wqjo5KiLZ-y+tBbXj7Lfapeybod5bpn9uW-5-re7gA@mail.gmail.com>

Our setup...

External UAC  192.168.x.x
  |
Router        5.x.x.x
  |
(internet)
  |
Firewall      46.x.x.x maps ports
  |           directly to
OpenSIPS      192.168.x.x      Mid-registrar and Mediaproxy
  |
Asterisk      192.168.x.x


Current situation:
- UAC can call destination registered on Asterisk on local n/w
- UAC registration "appears" to work via Mid-Registrar with AOR throttling
- Destination extension rings and can pick up call
- Audio works both ways (though drops after about 60 seconds)

However...

OPTIONS coming from Asterisk ends up being directed to the local IP address
at the UAC end (i.e. instead of going out over the internet it tries to use
the end destination local IP address following mid_registrar_lookup()).

Despite the UAC extension appearing to be registered on Asterisk, when I
call is placed to it, it immediately goes to voicemail - doesn't even seem
to attempt to call out to OpenSIPS - exactly as if no endpoint is
registered.


I've seen posts relating to this with earlier version of OpenSIPS and
setting mid_registrar insertion mode to 1, but that doesn't seem to be an
option with 3.1. What's the best way to handle this?
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From solarmon at one-n.co.uk  Mon Jan 18 16:04:19 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 18 Jan 2021 16:04:19 +0000
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
Message-ID: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>

Hi,,

I have a requirement to stop 183 without SDP packets from being passed, as
well as having to reply back with a PRACK.

I can stop the 183 without SDP from being passed on with the following in
onreply_route[]

 if ($rs == "183" && !has_body_part("application/sdp")) {
                drop();
                exit;
        }

However, how do I reply back with a PRACK?

Thank you.
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From osas at voipembedded.com  Mon Jan 18 16:09:27 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Mon, 18 Jan 2021 11:09:27 -0500
Subject: [OpenSIPS-Users] [SOLVED] 3.1 How to build DEB Debian Packages
 from source (Stretch)
In-Reply-To: <CALXCt0oFJB_s8hw_9w63yB54CD01N3iNJzP74b6POiAQANTH2g@mail.gmail.com>
References: <CALXCt0oCDhNxoLYPo1SAAarmzmEdvosQcOu1Jmq83LAhJzwy_w@mail.gmail.com>
 <CAND0LksiP5wC6s04otGO6eUNE+Y9do8fpMg7iz1Nuy8YOUZf7g@mail.gmail.com>
 <CALXCt0oFJB_s8hw_9w63yB54CD01N3iNJzP74b6POiAQANTH2g@mail.gmail.com>
Message-ID: <CAND0LksqORMD4c9+PmhoVikqaJh6gVDNVr4hyO0hkH50VYQBhg@mail.gmail.com>

Hello Giovanni,

All looks good!
IIRC you can alter the changelog to create custom names for packages (for
easier management).
Also, to simplify the install/update process, I create a custom mega
opensips package that has all the modules built in. That’s just my personal
preference :)

-ovidiu

On Mon, Jan 18, 2021 at 08:07 Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:

> My friends, Ovidiu,
>
> so, see the following two scripts, one for first time building deb
> packages, second script for updating such packages.
>
> Can you all please give hints, corrections, etc?
>
> FIRST TIME BUILD DEB PACKAGES:
> #=================
> apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
> dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
> libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
> libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
> libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
> libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
> pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
> libmongoc-dev
>
> mkdir /usr/src/OPENSIPS_DEBS
> cd /usr/src/OPENSIPS_DEBS
>
> git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
>
> cd opensips_3_1
>
> make deb-orig-tar
> make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"
>
> cd ..
> ls
>
> dpkg -i opensips_3.1.1-1_amd64.deb
> opensips-dialplan-module_3.1.1-1_amd64.deb
> opensips-http-modules_3.1.1-1_amd64.deb
> opensips-lua-module_3.1.1-1_amd64.deb
> opensips-memcached-module_3.1.1-1_amd64.deb
> opensips-postgres-module_3.1.1-1_amd64.deb
> opensips-presence-modules_3.1.1-1_amd64.deb
> opensips-redis-module_3.1.1-1_amd64.deb
> opensips-regex-module_3.1.1-1_amd64.deb
> opensips-restclient-module_3.1.1-1_amd64.deb
> opensips-sqlite-module_3.1.1-1_amd64.deb
> opensips-tlsmgm-module_3.1.1-1_amd64.deb
> opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
> opensips-xml-module_3.1.1-1_amd64.deb
> opensips-xmlrpc-module_3.1.1-1_amd64.deb
>
> #=================
>
> UPDATE DEB PACKAGES (keeping local source changes):
> #=================
> cd /usr/src/OPENSIPS_DEBS/opensips_3_1
>
> git stash ; git pull ; git stash apply
>
> make deb-orig-tar
> make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"
>
> cd ..
> ls
>
> dpkg -i opensips_3.1.1-1_amd64.deb
> opensips-dialplan-module_3.1.1-1_amd64.deb
> opensips-http-modules_3.1.1-1_amd64.deb
> opensips-lua-module_3.1.1-1_amd64.deb
> opensips-memcached-module_3.1.1-1_amd64.deb
> opensips-postgres-module_3.1.1-1_amd64.deb
> opensips-presence-modules_3.1.1-1_amd64.deb
> opensips-redis-module_3.1.1-1_amd64.deb
> opensips-regex-module_3.1.1-1_amd64.deb
> opensips-restclient-module_3.1.1-1_amd64.deb
> opensips-sqlite-module_3.1.1-1_amd64.deb
> opensips-tlsmgm-module_3.1.1-1_amd64.deb
> opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
> opensips-xml-module_3.1.1-1_amd64.deb
> opensips-xmlrpc-module_3.1.1-1_amd64.deb
>
> #=================
>
>
> On Sat, Jan 16, 2021 at 8:41 PM Ovidiu Sas <osas at voipembedded.com> wrote:
>
>> Hello Giovanni,
>>
>> After you clone the repo, you need to build the tar from it:
>> make deb-orig-tar
>> You can skip the wget.
>>
>> If you modify the source, rebuild the tar and then rebuild the deb
>> package.
>>
>> Also, when building the package, use the following options (so you
>> won't have troubles with signed packages):
>> make deb DEBBUILD_EXTRA_OPTIONS="-uc -us"
>>
>> -ovidiu
>>
>> On Sat, Jan 16, 2021 at 2:15 PM Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>> >
>> > Hello my friends,
>> >
>> > I know I'm slow, it took me much time to find the way to build deb
>> packages from source.
>> >
>> > So, without further ado, my receipt (on Stretch, may works on Buster
>> too)
>> > -BOTH git clone AND wget are required, it is not a typo- :
>> >
>> > =========
>> >
>> > mkdir /usr/src/OPENSIPS
>> > cd /usr/src/OPENSIPS
>> >
>> > git clone https://github.com/OpenSIPS/opensips.git -b 3.1 opensips_3_1
>> > wget https://github.com/OpenSIPS/opensips/archive/3.1.tar.gz -O
>> opensips_3.1.1.orig.tar.gz
>> >
>> > apt-get install bison debhelper default-libmysqlclient-dev dh-systemd
>> dpkg-dev  flex libconfuse-dev libcurl4-gnutls-dev libdb-dev libexpat1-dev
>> libmaxminddb-dev libhiredis-dev libjson-c-dev libldap2-dev liblua5.1-0-dev
>> libmemcached-dev libmicrohttpd-dev libbson-dev base-files libncurses5-dev
>> libpcre3-dev libperl-dev libpq-dev librabbitmq-dev libradcli-dev
>> libsctp-dev libsqlite3-dev libssl-dev lsb-release uuid-dev libxml2-dev
>> pkg-config python python-dev unixodbc-dev xsltproc zlib1g-dev libsnmp-dev
>> libmongoc-dev
>> >
>> > cd opensips_3_1
>> >
>> > make deb
>> > cd ..
>> >
>> > ls
>> >
>> > dpkg -i opensips_3.1.1-1_amd64.deb
>> opensips-dialplan-module_3.1.1-1_amd64.deb
>> opensips-http-modules_3.1.1-1_amd64.deb
>> opensips-lua-module_3.1.1-1_amd64.deb
>> opensips-memcached-module_3.1.1-1_amd64.deb
>> opensips-postgres-module_3.1.1-1_amd64.deb
>> opensips-presence-modules_3.1.1-1_amd64.deb
>> opensips-redis-module_3.1.1-1_amd64.deb
>> opensips-regex-module_3.1.1-1_amd64.deb
>> opensips-restclient-module_3.1.1-1_amd64.deb
>> opensips-sqlite-module_3.1.1-1_amd64.deb
>> opensips-tlsmgm-module_3.1.1-1_amd64.deb
>> opensips-tls-module_3.1.1-1_amd64.deb opensips-wss-module_3.1.1-1_amd64.deb
>> opensips-xml-module_3.1.1-1_amd64.deb
>> opensips-xmlrpc-module_3.1.1-1_amd64.deb
>> >
>> > =========
>> >
>> > Obviously, you can choose which deb packages to install :)
>> >
>> > HTH
>> >
>> > -giovanni
>> >
>> > --
>> > Sincerely,
>> >
>> > Giovanni Maruzzelli
>> > OpenTelecom.IT
>> > cell: +39 347 266 56 18
>> >
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.com
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
> --
VoIP Embedded, Inc.
http://www.voipembedded.com
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From Johan at democon.be  Mon Jan 18 16:10:33 2021
From: Johan at democon.be (Johan De Clercq)
Date: Mon, 18 Jan 2021 17:10:33 +0100
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
Message-ID: <CAEVWGm_4tdxgpH1ODtJ-3=xoq5VW4Oab-yV4sHMBB1p1uBkDsw@mail.gmail.com>

you mean that the remote party sends you required:100rel ? The only way
that I know is by using b2b logic.
Maybe Bogdan can shed some light here.

Op ma 18 jan. 2021 om 17:07 schreef solarmon <solarmon at one-n.co.uk>:

> Hi,,
>
> I have a requirement to stop 183 without SDP packets from being passed, as
> well as having to reply back with a PRACK.
>
> I can stop the 183 without SDP from being passed on with the following in
> onreply_route[]
>
>  if ($rs == "183" && !has_body_part("application/sdp")) {
>                 drop();
>                 exit;
>         }
>
> However, how do I reply back with a PRACK?
>
> Thank you.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From solarmon at one-n.co.uk  Mon Jan 18 16:28:35 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 18 Jan 2021 16:28:35 +0000
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAEVWGm_4tdxgpH1ODtJ-3=xoq5VW4Oab-yV4sHMBB1p1uBkDsw@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
 <CAEVWGm_4tdxgpH1ODtJ-3=xoq5VW4Oab-yV4sHMBB1p1uBkDsw@mail.gmail.com>
Message-ID: <CAOXfywNXYa1RKsBfQzMJ1kwmXb9jtqzYuwoYui0fEaq3xd3oDA@mail.gmail.com>

Hi Johan,

The first 183 packet that is received has no SDP in it and has:

Require: 100rel

I wish to prevent this from being passed on, but still respond what with a
PRACK (since 100rel has been requested). The reason for this is that our
system will not work if it first receives a 183 without SDP and then
receives a second 183 with SDP.

Thank you.

On Mon, 18 Jan 2021 at 16:13, Johan De Clercq <Johan at democon.be> wrote:

> you mean that the remote party sends you required:100rel ? The only way
> that I know is by using b2b logic.
> Maybe Bogdan can shed some light here.
>
> Op ma 18 jan. 2021 om 17:07 schreef solarmon <solarmon at one-n.co.uk>:
>
>> Hi,,
>>
>> I have a requirement to stop 183 without SDP packets from being passed,
>> as well as having to reply back with a PRACK.
>>
>> I can stop the 183 without SDP from being passed on with the following in
>> onreply_route[]
>>
>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>                 drop();
>>                 exit;
>>         }
>>
>> However, how do I reply back with a PRACK?
>>
>> Thank you.
>> _______________________________________________
>> 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 tpaivaa at gmail.com  Mon Jan 18 17:32:34 2021
From: tpaivaa at gmail.com (Tomi Hakkarainen)
Date: Mon, 18 Jan 2021 19:32:34 +0200
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAOXfywNXYa1RKsBfQzMJ1kwmXb9jtqzYuwoYui0fEaq3xd3oDA@mail.gmail.com>
References: <CAOXfywNXYa1RKsBfQzMJ1kwmXb9jtqzYuwoYui0fEaq3xd3oDA@mail.gmail.com>
Message-ID: <C340454F-DC49-47BB-877F-CCD766370EEE@gmail.com>

Hi,

I guess does not answer the guestion but take a look at this:

https://github.com/OpenSIPS/opensips/issues/2076

that said could you setup b2b logic to fix interworking issues?

Tomi

On 18. Jan 2021, at 18.31, solarmon <solarmon at one-n.co.uk> wrote:


Hi Johan,

The first 183 packet that is received has no SDP in it and has:

Require: 100rel

I wish to prevent this from being passed on, but still respond what with a PRACK (since 100rel has been requested). The reason for this is that our system will not work if it first receives a 183 without SDP and then receives a second 183 with SDP.

Thank you.

On Mon, 18 Jan 2021 at 16:13, Johan De Clercq <Johan at democon.be> wrote:
> you mean that the remote party sends you required:100rel ? The only way that I know is by using b2b logic. 
> Maybe Bogdan can shed some light here. 
> 
> Op ma 18 jan. 2021 om 17:07 schreef solarmon <solarmon at one-n.co.uk>:
>> Hi,,
>> 
>> I have a requirement to stop 183 without SDP packets from being passed, as well as having to reply back with a PRACK.
>> 
>> I can stop the 183 without SDP from being passed on with the following in onreply_route[]
>> 
>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>                 drop();
>>                 exit;
>>         }
>> 
>> However, how do I reply back with a PRACK?
>> 
>> Thank you.
>> _______________________________________________
>> 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
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From solarmon at one-n.co.uk  Mon Jan 18 18:33:05 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 18 Jan 2021 18:33:05 +0000
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <C340454F-DC49-47BB-877F-CCD766370EEE@gmail.com>
References: <CAOXfywNXYa1RKsBfQzMJ1kwmXb9jtqzYuwoYui0fEaq3xd3oDA@mail.gmail.com>
 <C340454F-DC49-47BB-877F-CCD766370EEE@gmail.com>
Message-ID: <CAOXfywP-jZoSZm0DvGHhhNM9Qu=c4Zt1tbT4izj=mexpOQOe8Q@mail.gmail.com>

Hi Toni,

Thanks, I did come across that.

I was hoping not to have to install a module and have to do even more
changes to accommodate it. I was hoping that there was already a core
feature/mechanism to handle responding back with a PRACK.

Thank you.

On Mon, 18 Jan 2021, 17:35 Tomi Hakkarainen, <tpaivaa at gmail.com> wrote:

> Hi,
>
> I guess does not answer the guestion but take a look at this:
>
> https://github.com/OpenSIPS/opensips/issues/2076
>
> that said could you setup b2b logic to fix interworking issues?
>
> Tomi
>
> On 18. Jan 2021, at 18.31, solarmon <solarmon at one-n.co.uk> wrote:
>
> 
> Hi Johan,
>
> The first 183 packet that is received has no SDP in it and has:
>
> Require: 100rel
>
> I wish to prevent this from being passed on, but still respond what with a
> PRACK (since 100rel has been requested). The reason for this is that our
> system will not work if it first receives a 183 without SDP and then
> receives a second 183 with SDP.
>
> Thank you.
>
> On Mon, 18 Jan 2021 at 16:13, Johan De Clercq <Johan at democon.be> wrote:
>
>> you mean that the remote party sends you required:100rel ? The only way
>> that I know is by using b2b logic.
>> Maybe Bogdan can shed some light here.
>>
>> Op ma 18 jan. 2021 om 17:07 schreef solarmon <solarmon at one-n.co.uk>:
>>
>>> Hi,,
>>>
>>> I have a requirement to stop 183 without SDP packets from being passed,
>>> as well as having to reply back with a PRACK.
>>>
>>> I can stop the 183 without SDP from being passed on with the following
>>> in onreply_route[]
>>>
>>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>>                 drop();
>>>                 exit;
>>>         }
>>>
>>> However, how do I reply back with a PRACK?
>>>
>>> Thank you.
>>> _______________________________________________
>>> 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
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From alex at alexkinch.com  Mon Jan 18 18:53:21 2021
From: alex at alexkinch.com (Alex Kinch)
Date: Mon, 18 Jan 2021 18:53:21 +0000
Subject: [OpenSIPS-Users] Global variable $rm gives number when using $json
Message-ID: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>

Hi all,

I'm trying to post a few variables in JSON in OpenSIPS 3.1, but $rm is
returning a number instead of the method in the JSON.

If I do something like this:

$json(body) := "{}";
$json(body/call_id) = $ci;
$json(body/ts) = $time(%Y-%m-%d %H:%M:%SZ);
$json(body/src_ip) = $si;
$json(body/dst_ip) = $socket_in(ip);
$json(body/method) = $rm;
$json(body/sip_from) = $fu;
$json(body/sip_to) = $tu;
$json(body/dialled) = $tU;
xlog("L_NOTICE", "request: $rm from $fu to $ru\n");
xlog("L_NOTICE", "Sending $json(body)");

I get this (identifying data redacted):

Jan 18 18:18:15 [33] request: INVITE from sip:XX at XX to sip:XX at XX
Jan 18 18:18:15 [33] Sending { "call_id": "942887463-1604195939-951239259",
"ts": "2021-01-18 18:18:15Z", "src_ip": "XX", "dst_ip": "XX", "method": 1,
"sip_from": "sip:XX at XX", "sip_to": "sip:XX at XX", "dialled": "XX" }

Any suggestions?

Thanks
Alex
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From osas at voipembedded.com  Mon Jan 18 19:01:33 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Mon, 18 Jan 2021 14:01:33 -0500
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
Message-ID: <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>

You can change the 183 into an 180 and let the prack take its course.

Alternatively, you can remove 100rel from the initial INVITE and drop the
183 without SDP.

-ovidiu

On Mon, Jan 18, 2021 at 11:05 solarmon <solarmon at one-n.co.uk> wrote:

> Hi,,
>
> I have a requirement to stop 183 without SDP packets from being passed, as
> well as having to reply back with a PRACK.
>
> I can stop the 183 without SDP from being passed on with the following in
> onreply_route[]
>
>  if ($rs == "183" && !has_body_part("application/sdp")) {
>                 drop();
>                 exit;
>         }
>
> However, how do I reply back with a PRACK?
>
> Thank you.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
VoIP Embedded, Inc.
http://www.voipembedded.com
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From kworm at missouri-telecom.com  Mon Jan 18 19:29:11 2021
From: kworm at missouri-telecom.com (Kevin Wormington)
Date: Mon, 18 Jan 2021 13:29:11 -0600
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on node
	failure
Message-ID: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>

Hi,

I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.  

What I have working so far:

Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.

What I can’t get to work:

Calls that are already in progress to switch between nodes when one node fails.


I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?

Thanks,

Kevin

From solarmon at one-n.co.uk  Mon Jan 18 20:11:51 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 18 Jan 2021 20:11:51 +0000
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
 <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>
Message-ID: <CAOXfywPdUt5ofo-=OjeDjbwEWHnhxBcgS=qYXrK=C=S8L4K5Ew@mail.gmail.com>

H Ovidiu

We do not want our system to change to a 'ringing' state, which would
happen if it receives 180. We would like to drop the first 183 (without
SDP) but reply back with an PRACK (since 100rel was sent) and let the
subsequent 183 (with SDP) through. This is just a quirk in our system that
we are looking to find a workaround for.

I don't think removing 100rel from the INVITE would work? The sender of the
INVITE has put in 100rel and is expecting a PRACK, so us removing it does
not change this expectation?

Thank you.

On Mon, 18 Jan 2021 at 19:03, Ovidiu Sas <osas at voipembedded.com> wrote:

> You can change the 183 into an 180 and let the prack take its course.
>
> Alternatively, you can remove 100rel from the initial INVITE and drop the
> 183 without SDP.
>
> -ovidiu
>
> On Mon, Jan 18, 2021 at 11:05 solarmon <solarmon at one-n.co.uk> wrote:
>
>> Hi,,
>>
>> I have a requirement to stop 183 without SDP packets from being passed,
>> as well as having to reply back with a PRACK.
>>
>> I can stop the 183 without SDP from being passed on with the following in
>> onreply_route[]
>>
>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>                 drop();
>>                 exit;
>>         }
>>
>> However, how do I reply back with a PRACK?
>>
>> Thank you.
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From osas at voipembedded.com  Mon Jan 18 21:51:18 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Mon, 18 Jan 2021 16:51:18 -0500
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAOXfywPdUt5ofo-=OjeDjbwEWHnhxBcgS=qYXrK=C=S8L4K5Ew@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
 <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>
 <CAOXfywPdUt5ofo-=OjeDjbwEWHnhxBcgS=qYXrK=C=S8L4K5Ew@mail.gmail.com>
Message-ID: <CAND0LkvvC8pa_NxQPPuKDABPea8JoOpxGpPVOgQQgFE4_5uoLg@mail.gmail.com>

100rel in the initial INVITE means that the caller has support for it, if
requested by the caller. If not present in INVITE, the caller should not
send provisional replies with 100rel.

-ovidiu

On Mon, Jan 18, 2021 at 15:13 solarmon <solarmon at one-n.co.uk> wrote:

> H Ovidiu
>
> We do not want our system to change to a 'ringing' state, which would
> happen if it receives 180. We would like to drop the first 183 (without
> SDP) but reply back with an PRACK (since 100rel was sent) and let the
> subsequent 183 (with SDP) through. This is just a quirk in our system that
> we are looking to find a workaround for.
>
> I don't think removing 100rel from the INVITE would work? The sender of
> the INVITE has put in 100rel and is expecting a PRACK, so us removing it
> does not change this expectation?
>
> Thank you.
>
> On Mon, 18 Jan 2021 at 19:03, Ovidiu Sas <osas at voipembedded.com> wrote:
>
>> You can change the 183 into an 180 and let the prack take its course.
>>
>> Alternatively, you can remove 100rel from the initial INVITE and drop the
>> 183 without SDP.
>>
>> -ovidiu
>>
>> On Mon, Jan 18, 2021 at 11:05 solarmon <solarmon at one-n.co.uk> wrote:
>>
>>> Hi,,
>>>
>>> I have a requirement to stop 183 without SDP packets from being passed,
>>> as well as having to reply back with a PRACK.
>>>
>>> I can stop the 183 without SDP from being passed on with the following
>>> in onreply_route[]
>>>
>>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>>                 drop();
>>>                 exit;
>>>         }
>>>
>>> However, how do I reply back with a PRACK?
>>>
>>> Thank you.
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.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
>
-- 
VoIP Embedded, Inc.
http://www.voipembedded.com
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From solarmon at one-n.co.uk  Mon Jan 18 22:07:09 2021
From: solarmon at one-n.co.uk (solarmon)
Date: Mon, 18 Jan 2021 22:07:09 +0000
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAND0LkvvC8pa_NxQPPuKDABPea8JoOpxGpPVOgQQgFE4_5uoLg@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
 <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>
 <CAOXfywPdUt5ofo-=OjeDjbwEWHnhxBcgS=qYXrK=C=S8L4K5Ew@mail.gmail.com>
 <CAND0LkvvC8pa_NxQPPuKDABPea8JoOpxGpPVOgQQgFE4_5uoLg@mail.gmail.com>
Message-ID: <CAOXfywOYj+D13BA4mOAXEL8d_=xDMU3aQEbPdcAar1cV1R2Ozg@mail.gmail.com>

The call flow is that we (our system) are the caller - we send the first
INVITE. The SIP provider is responding back with 183s because they want to
initiate early media.

Are you suggesting that we remove 100rel from our INVITE that we send out
to the SIP provider?

On Mon, 18 Jan 2021 at 21:54, Ovidiu Sas <osas at voipembedded.com> wrote:

> 100rel in the initial INVITE means that the caller has support for it, if
> requested by the caller. If not present in INVITE, the caller should not
> send provisional replies with 100rel.
>
> -ovidiu
>
> On Mon, Jan 18, 2021 at 15:13 solarmon <solarmon at one-n.co.uk> wrote:
>
>> H Ovidiu
>>
>> We do not want our system to change to a 'ringing' state, which would
>> happen if it receives 180. We would like to drop the first 183 (without
>> SDP) but reply back with an PRACK (since 100rel was sent) and let the
>> subsequent 183 (with SDP) through. This is just a quirk in our system that
>> we are looking to find a workaround for.
>>
>> I don't think removing 100rel from the INVITE would work? The sender of
>> the INVITE has put in 100rel and is expecting a PRACK, so us removing it
>> does not change this expectation?
>>
>> Thank you.
>>
>> On Mon, 18 Jan 2021 at 19:03, Ovidiu Sas <osas at voipembedded.com> wrote:
>>
>>> You can change the 183 into an 180 and let the prack take its course.
>>>
>>> Alternatively, you can remove 100rel from the initial INVITE and drop
>>> the 183 without SDP.
>>>
>>> -ovidiu
>>>
>>> On Mon, Jan 18, 2021 at 11:05 solarmon <solarmon at one-n.co.uk> wrote:
>>>
>>>> Hi,,
>>>>
>>>> I have a requirement to stop 183 without SDP packets from being passed,
>>>> as well as having to reply back with a PRACK.
>>>>
>>>> I can stop the 183 without SDP from being passed on with the following
>>>> in onreply_route[]
>>>>
>>>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>>>                 drop();
>>>>                 exit;
>>>>         }
>>>>
>>>> However, how do I reply back with a PRACK?
>>>>
>>>> Thank you.
>>>> _______________________________________________
>>>> Users mailing list
>>>> Users at lists.opensips.org
>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>> --
>>> VoIP Embedded, Inc.
>>> http://www.voipembedded.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
>>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From osas at voipembedded.com  Mon Jan 18 22:30:44 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Mon, 18 Jan 2021 17:30:44 -0500
Subject: [OpenSIPS-Users] PRACK response to 183 without SDP
In-Reply-To: <CAOXfywOYj+D13BA4mOAXEL8d_=xDMU3aQEbPdcAar1cV1R2Ozg@mail.gmail.com>
References: <CAOXfywNMdEpNHrqsRfe7fjixVP5JSJgP0C8CybKGK+DEZ6OtAA@mail.gmail.com>
 <CAND0LktaT+yW8weTP7iiu4kJacH2ELS-_-yNkOHtc7h0yJKhzA@mail.gmail.com>
 <CAOXfywPdUt5ofo-=OjeDjbwEWHnhxBcgS=qYXrK=C=S8L4K5Ew@mail.gmail.com>
 <CAND0LkvvC8pa_NxQPPuKDABPea8JoOpxGpPVOgQQgFE4_5uoLg@mail.gmail.com>
 <CAOXfywOYj+D13BA4mOAXEL8d_=xDMU3aQEbPdcAar1cV1R2Ozg@mail.gmail.com>
Message-ID: <CAND0LktXBK9WzhUcUr2CnoKmvmdzO=N8mc02XA6fvDiKPKZiAg@mail.gmail.com>

That's exactly what I suggest! If you remove 100rel from INVITE, the
SIP provider should send a regular 183, not a 183 with 100rel. Then
you can safely drop the 183 without SDP.

-ovidiu

On Mon, Jan 18, 2021 at 5:08 PM solarmon <solarmon at one-n.co.uk> wrote:
>
> The call flow is that we (our system) are the caller - we send the first INVITE. The SIP provider is responding back with 183s because they want to initiate early media.
>
> Are you suggesting that we remove 100rel from our INVITE that we send out to the SIP provider?
>
> On Mon, 18 Jan 2021 at 21:54, Ovidiu Sas <osas at voipembedded.com> wrote:
>>
>> 100rel in the initial INVITE means that the caller has support for it, if requested by the caller. If not present in INVITE, the caller should not send provisional replies with 100rel.
>>
>> -ovidiu
>>
>> On Mon, Jan 18, 2021 at 15:13 solarmon <solarmon at one-n.co.uk> wrote:
>>>
>>> H Ovidiu
>>>
>>> We do not want our system to change to a 'ringing' state, which would happen if it receives 180. We would like to drop the first 183 (without SDP) but reply back with an PRACK (since 100rel was sent) and let the subsequent 183 (with SDP) through. This is just a quirk in our system that we are looking to find a workaround for.
>>>
>>> I don't think removing 100rel from the INVITE would work? The sender of the INVITE has put in 100rel and is expecting a PRACK, so us removing it does not change this expectation?
>>>
>>> Thank you.
>>>
>>> On Mon, 18 Jan 2021 at 19:03, Ovidiu Sas <osas at voipembedded.com> wrote:
>>>>
>>>> You can change the 183 into an 180 and let the prack take its course.
>>>>
>>>> Alternatively, you can remove 100rel from the initial INVITE and drop the 183 without SDP.
>>>>
>>>> -ovidiu
>>>>
>>>> On Mon, Jan 18, 2021 at 11:05 solarmon <solarmon at one-n.co.uk> wrote:
>>>>>
>>>>> Hi,,
>>>>>
>>>>> I have a requirement to stop 183 without SDP packets from being passed, as well as having to reply back with a PRACK.
>>>>>
>>>>> I can stop the 183 without SDP from being passed on with the following in onreply_route[]
>>>>>
>>>>>  if ($rs == "183" && !has_body_part("application/sdp")) {
>>>>>                 drop();
>>>>>                 exit;
>>>>>         }
>>>>>
>>>>> However, how do I reply back with a PRACK?
>>>>>
>>>>> Thank you.
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> VoIP Embedded, Inc.
>>>> http://www.voipembedded.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
>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.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



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com


From conorjpower at hotmail.com  Wed Jan 13 23:42:58 2021
From: conorjpower at hotmail.com (Conor Power)
Date: Wed, 13 Jan 2021 23:42:58 +0000
Subject: [OpenSIPS-Users] OpenSIPS as simple SIP proxy
In-Reply-To: <CAEVWGm-zhzcCo2-+aUWRjhu251pWhbCNcdEwrviRUevgx0Q1hw@mail.gmail.com>
References: <BYAPR06MB554173BAFBF1B18DC2BA552AA5A90@BYAPR06MB5541.namprd06.prod.outlook.com>,
 <CAEVWGm-zhzcCo2-+aUWRjhu251pWhbCNcdEwrviRUevgx0Q1hw@mail.gmail.com>
Message-ID: <BYAPR06MB5541A5BCBF35E35718941789A5A90@BYAPR06MB5541.namprd06.prod.outlook.com>

Thanks Johan.

I generated the residential config and am using that now. It looks as follows:


route{


        if (!mf_process_maxfwd_header(10)) {

                send_reply(483,"Too Many Hops");

                exit;

        }


        if (has_totag()) {


                # handle hop-by-hop ACK (no routing required)

                if ( is_method("ACK") && t_check_trans() ) {

                        t_relay();

                        exit;

                }


                # sequential request within a dialog should

                # take the path determined by record-routing

                if ( !loose_route() ) {

                        # we do record-routing for all our traffic, so we should not

                        # receive any sequential requests without Route hdr.

                        send_reply(404,"Not here");

                        exit;

                }


                if (is_method("BYE")) {

                        # do accounting even if the transaction fails

                        do_accounting("log","failed");

                }


                # route it out to whatever destination was set by loose_route()

                # in $du (destination URI).

                route(relay);

                exit;

        }


        # CANCEL processing

        if (is_method("CANCEL")) {

                if (t_check_trans())

                        t_relay();

                exit;

        }


        # absorb retransmissions, but do not create transaction

        t_check_trans();


        if ( !(is_method("REGISTER")  ) ) {


                if (is_myself("$fd")) {


                } else {

                        # if caller is not local, then called number must be local


                        if (!is_myself("$rd")) {

                                send_reply(403,"Relay Forbidden");

                                exit;

                        }

                }


        }


        # preloaded route checking

        if (loose_route()) {

                xlog("L_ERR",

                        "Attempt to route with preloaded Route's [$fu/$tu/$ru/$ci]");

                if (!is_method("ACK"))

                        send_reply(403,"Preload Route denied");

                exit;

        }


        # record routing

        if (!is_method("REGISTER|MESSAGE"))

                record_route();


        # account only INVITEs

        if (is_method("INVITE")) {


                do_accounting("log");

        }



        if (!is_myself("$rd")) {

                append_hf("P-hint: outbound\r\n");


                route(relay);

        }


        # requests for my domain


        if (is_method("PUBLISH|SUBSCRIBE")) {

                send_reply(503, "Service Unavailable");

                exit;

        }


        if (is_method("REGISTER")) {


                if ($socket_in(proto) == "tcp")

                        setflag("TCP_PERSISTENT");


                if (!save("location"))

                        sl_reply_error();


                exit;

        }


        if ($rU==NULL) {

                # request with no Username in RURI

                send_reply(484,"Address Incomplete");

                exit;

        }








        # do lookup with method filtering

        if (!lookup("location","m")) {



                t_reply(404, "Not Found");

                exit;

        }




        # when routing via usrloc, log the missed calls also

        do_accounting("log","missed");

        route(relay);

}

I'm not sure where I should be adding the sethostport in this config.

Any further pointers would be great or to a relevant mail or tutorial.

Conor
________________________________
From: Users <users-bounces at lists.opensips.org> on behalf of Johan De Clercq <Johan at democon.be>
Sent: Wednesday, January 13, 2021 12:27 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS as simple SIP proxy

take the residential config and see where that gets you.

Op wo 13 jan. 2021 om 08:21 schreef Conor Power <conorjpower at hotmail.com<mailto:conorjpower at hotmail.com>>:
Hi,
   Apologies for the noob question but I'm hoping someone can point me in the right direction.

I am trying to use OpenSIPS as a simple proxy to proxy all calls to another SIP endpoint and back again to the original client. The only role of the OpenSIPS server is to function as the proxy and it is for all requests.

I have OpenSIPS up and running and can see the requests coming inbound using ngrep but I've had no success proxying the requests.

I added a sethostport() call in the config file but really am not sure where or how it fits in.

If someone might point me to a simple config file that would be used for such a proxy setup, it would be greatly appreciated.

Conor
_______________________________________________
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 volga629 at skillsearch.ca  Thu Jan 14 00:17:38 2021
From: volga629 at skillsearch.ca (volga629)
Date: Wed, 13 Jan 2021 20:17:38 -0400
Subject: [OpenSIPS-Users] sangoma issue ?
In-Reply-To: <CAKe=zzgkPaixKs=wz5VECUTVUm0fhuuhPtBXp4VQDuS=pOK-rg@mail.gmail.com>
References: <CAKe=zzhL8odExBoKSa1WG6erX7z8fo0GuCvAsCn060s64+3_ig@mail.gmail.com>
 <CAKe=zzhgq2vSeqgJeRtzjtn9yRPvf2C+Xd=ez3KoQbhWchaQpQ@mail.gmail.com>
 <PR1PR06MB46837F7B3E7053F867CFAF49A7A90@PR1PR06MB4683.eurprd06.prod.outlook.com>
 <CAKe=zzgkPaixKs=wz5VECUTVUm0fhuuhPtBXp4VQDuS=pOK-rg@mail.gmail.com>
Message-ID: <217c6ce2-d065-6209-b8b0-706bc18a745d@skillsearch.ca>

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From adierlam at ptgi-ics.com  Tue Jan 19 14:46:37 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Tue, 19 Jan 2021 09:46:37 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
	node failure
In-Reply-To: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
Message-ID: <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>

With dialog writing to db that both servers use.   And same tag on both -
modparam("dialog", "dlg_sharing_tag", "vip1=active")
had this working on opensips 2.4

thanks
Andy


On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com>
wrote:

> Hi,
>
> I've been attempting to get a two node active/active setup to work with
> the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted
> by a proxy that handles all of the NAT/media so either OpenSIPS instance
> can communicate directly with the user.
>
> What I have working so far:
>
> Registrations and calls work when sent to either node and if you stop
> OpenSIPS on a node new calls work fine using the other node.
>
> What I can’t get to work:
>
> Calls that are already in progress to switch between nodes when one node
> fails.
>
>
> I have messed around with various sharing tags…no tag, same tag, different
> tags but haven’t had any luck.   I’m guessing that I’m missing something to
> trigger the remaining node to send re-invites.  Has anyone attempted this
> type of setup and have any ideas?
>
> Thanks,
>
> Kevin
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From kworm at missouri-telecom.com  Tue Jan 19 15:00:17 2021
From: kworm at missouri-telecom.com (Kevin Wormington)
Date: Tue, 19 Jan 2021 09:00:17 -0600
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
Message-ID: <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>

I’m not using a VIP and I have made some progress by setting a different active tag on each node…then upon node failure setting the failed node's tag to active on remaining node.  This lets the re-invite pinging work, etc.  It’s almost there but the handling of the BYE…they are still sent to the IP of the failed node even after re-invite pings so any in-progress calls from the failed node are zombie when they hang up until the re-invite ping times out (30 seconds).   I found an article about initiating a re-invite on the new node with something like "opensips-cli -x mi dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but that either seems to terminate the call immediately or say the dialog wasn’t found.


Thanks,

Kevin
> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> 
> With dialog writing to db that both servers use.   And same tag on both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> had this working on opensips 2.4 
> 
> thanks
> Andy
> 
> 
> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com> wrote:
> Hi,
> 
> I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.  
> 
> What I have working so far:
> 
> Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.
> 
> What I can’t get to work:
> 
> Calls that are already in progress to switch between nodes when one node fails.
> 
> 
> I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?
> 
> Thanks,
> 
> Kevin
> _______________________________________________
> 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 adierlam at ptgi-ics.com  Tue Jan 19 15:08:39 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Tue, 19 Jan 2021 10:08:39 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
	node failure
In-Reply-To: <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
Message-ID: <CAKe=zzimp-NW7zUyoP5N5=Mu9pwN-DdWhTB6d23BZDFLdwmppQ@mail.gmail.com>

Ah, my setup was with a floating IP between servers.

thanks
Andy

On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington <
kworm at missouri-telecom.com> wrote:

> I’m not using a VIP and I have made some progress by setting a different
> active tag on each node…then upon node failure setting the failed node's
> tag to active on remaining node.  This lets the re-invite pinging work,
> etc.  It’s almost there but the handling of the BYE…they are still sent to
> the IP of the failed node even after re-invite pings so any in-progress
> calls from the failed node are zombie when they hang up until the re-invite
> ping times out (30 seconds).   I found an article about initiating a
> re-invite on the new node with something like "opensips-cli -x mi
> dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700"
> mode=challenge body=outbound” but that either seems to terminate the call
> immediately or say the dialog wasn’t found.
>
>
> Thanks,
>
> Kevin
> > On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> >
> > With dialog writing to db that both servers use.   And same tag on both
> - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > had this working on opensips 2.4
> >
> > thanks
> > Andy
> >
> >
> > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <
> kworm at missouri-telecom.com> wrote:
> > Hi,
> >
> > I've been attempting to get a two node active/active setup to work with
> the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted
> by a proxy that handles all of the NAT/media so either OpenSIPS instance
> can communicate directly with the user.
> >
> > What I have working so far:
> >
> > Registrations and calls work when sent to either node and if you stop
> OpenSIPS on a node new calls work fine using the other node.
> >
> > What I can’t get to work:
> >
> > Calls that are already in progress to switch between nodes when one node
> fails.
> >
> >
> > I have messed around with various sharing tags…no tag, same tag,
> different tags but haven’t had any luck.   I’m guessing that I’m missing
> something to trigger the remaining node to send re-invites.  Has anyone
> attempted this type of setup and have any ideas?
> >
> > Thanks,
> >
> > Kevin
> > _______________________________________________
> > 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
>
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From kworm at missouri-telecom.com  Tue Jan 19 15:19:47 2021
From: kworm at missouri-telecom.com (Kevin Wormington)
Date: Tue, 19 Jan 2021 09:19:47 -0600
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <CAKe=zzimp-NW7zUyoP5N5=Mu9pwN-DdWhTB6d23BZDFLdwmppQ@mail.gmail.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
 <CAKe=zzimp-NW7zUyoP5N5=Mu9pwN-DdWhTB6d23BZDFLdwmppQ@mail.gmail.com>
Message-ID: <3EA584D5-A47D-4E35-9D4B-5147BC875287@missouri-telecom.com>

That seems to be how most are setup…maybe I’m making it harder than it should be :-)

Out of curiosity what did/do you use to monitor OpenSIPS as up for your failover or did you just rely on the IP (keepalived, etc.) reachability? 

Thanks,

Kevin
> On Jan 19, 2021, at 9:08 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> 
> Ah, my setup was with a floating IP between servers.     
> 
> thanks
> Andy
> 
> On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington <kworm at missouri-telecom.com> wrote:
> I’m not using a VIP and I have made some progress by setting a different active tag on each node…then upon node failure setting the failed node's tag to active on remaining node.  This lets the re-invite pinging work, etc.  It’s almost there but the handling of the BYE…they are still sent to the IP of the failed node even after re-invite pings so any in-progress calls from the failed node are zombie when they hang up until the re-invite ping times out (30 seconds).   I found an article about initiating a re-invite on the new node with something like "opensips-cli -x mi dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but that either seems to terminate the call immediately or say the dialog wasn’t found.
> 
> 
> Thanks,
> 
> Kevin
> > On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> > 
> > With dialog writing to db that both servers use.   And same tag on both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > had this working on opensips 2.4 
> > 
> > thanks
> > Andy
> > 
> > 
> > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com> wrote:
> > Hi,
> > 
> > I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.  
> > 
> > What I have working so far:
> > 
> > Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.
> > 
> > What I can’t get to work:
> > 
> > Calls that are already in progress to switch between nodes when one node fails.
> > 
> > 
> > I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?
> > 
> > Thanks,
> > 
> > Kevin
> > _______________________________________________
> > 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
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From adierlam at ptgi-ics.com  Tue Jan 19 15:26:46 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Tue, 19 Jan 2021 10:26:46 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
	node failure
In-Reply-To: <3EA584D5-A47D-4E35-9D4B-5147BC875287@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
 <CAKe=zzimp-NW7zUyoP5N5=Mu9pwN-DdWhTB6d23BZDFLdwmppQ@mail.gmail.com>
 <3EA584D5-A47D-4E35-9D4B-5147BC875287@missouri-telecom.com>
Message-ID: <CAKe=zzjAKFi7YVrkqnvc+myfb0vMtGHYx+sXjNQqwN_=hSmJxw@mail.gmail.com>

*Still in the process of testing/learning.  Would trigger the fail-over by
shutting down keepalived.

thanks,
Andy



On Tue, Jan 19, 2021 at 10:22 AM Kevin Wormington <
kworm at missouri-telecom.com> wrote:

> That seems to be how most are setup…maybe I’m making it harder than it
> should be :-)
>
> Out of curiosity what did/do you use to monitor OpenSIPS as up for your
> failover or did you just rely on the IP (keepalived, etc.) reachability?
>
> Thanks,
>
> Kevin
> > On Jan 19, 2021, at 9:08 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> >
> > Ah, my setup was with a floating IP between servers.
> >
> > thanks
> > Andy
> >
> > On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington <
> kworm at missouri-telecom.com> wrote:
> > I’m not using a VIP and I have made some progress by setting a different
> active tag on each node…then upon node failure setting the failed node's
> tag to active on remaining node.  This lets the re-invite pinging work,
> etc.  It’s almost there but the handling of the BYE…they are still sent to
> the IP of the failed node even after re-invite pings so any in-progress
> calls from the failed node are zombie when they hang up until the re-invite
> ping times out (30 seconds).   I found an article about initiating a
> re-invite on the new node with something like "opensips-cli -x mi
> dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700"
> mode=challenge body=outbound” but that either seems to terminate the call
> immediately or say the dialog wasn’t found.
> >
> >
> > Thanks,
> >
> > Kevin
> > > On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com>
> wrote:
> > >
> > > With dialog writing to db that both servers use.   And same tag on
> both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > > had this working on opensips 2.4
> > >
> > > thanks
> > > Andy
> > >
> > >
> > > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <
> kworm at missouri-telecom.com> wrote:
> > > Hi,
> > >
> > > I've been attempting to get a two node active/active setup to work
> with the v3.1 clusterer module sharing usrloc and dialog.  The setup is
> fronted by a proxy that handles all of the NAT/media so either OpenSIPS
> instance can communicate directly with the user.
> > >
> > > What I have working so far:
> > >
> > > Registrations and calls work when sent to either node and if you stop
> OpenSIPS on a node new calls work fine using the other node.
> > >
> > > What I can’t get to work:
> > >
> > > Calls that are already in progress to switch between nodes when one
> node fails.
> > >
> > >
> > > I have messed around with various sharing tags…no tag, same tag,
> different tags but haven’t had any luck.   I’m guessing that I’m missing
> something to trigger the remaining node to send re-invites.  Has anyone
> attempted this type of setup and have any ideas?
> > >
> > > Thanks,
> > >
> > > Kevin
> > > _______________________________________________
> > > 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
> > _______________________________________________
> > 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  Tue Jan 19 15:28:31 2021
From: Johan at democon.be (Johan De Clercq)
Date: Tue, 19 Jan 2021 16:28:31 +0100
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
	node failure
In-Reply-To: <3EA584D5-A47D-4E35-9D4B-5147BC875287@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
 <CAKe=zzimp-NW7zUyoP5N5=Mu9pwN-DdWhTB6d23BZDFLdwmppQ@mail.gmail.com>
 <3EA584D5-A47D-4E35-9D4B-5147BC875287@missouri-telecom.com>
Message-ID: <CAEVWGm8MdOd2M9wuU2UP3Qp=2xvKEh6ZE0CX+Xz=Gtj_wHe-KQ@mail.gmail.com>

Same here. Floating ip

On Tue, Jan 19, 2021, 16:23 Kevin Wormington <kworm at missouri-telecom.com>
wrote:

> That seems to be how most are setup…maybe I’m making it harder than it
> should be :-)
>
> Out of curiosity what did/do you use to monitor OpenSIPS as up for your
> failover or did you just rely on the IP (keepalived, etc.) reachability?
>
> Thanks,
>
> Kevin
> > On Jan 19, 2021, at 9:08 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
> >
> > Ah, my setup was with a floating IP between servers.
> >
> > thanks
> > Andy
> >
> > On Tue, Jan 19, 2021 at 10:02 AM Kevin Wormington <
> kworm at missouri-telecom.com> wrote:
> > I’m not using a VIP and I have made some progress by setting a different
> active tag on each node…then upon node failure setting the failed node's
> tag to active on remaining node.  This lets the re-invite pinging work,
> etc.  It’s almost there but the handling of the BYE…they are still sent to
> the IP of the failed node even after re-invite pings so any in-progress
> calls from the failed node are zombie when they hang up until the re-invite
> ping times out (30 seconds).   I found an article about initiating a
> re-invite on the new node with something like "opensips-cli -x mi
> dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700"
> mode=challenge body=outbound” but that either seems to terminate the call
> immediately or say the dialog wasn’t found.
> >
> >
> > Thanks,
> >
> > Kevin
> > > On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com>
> wrote:
> > >
> > > With dialog writing to db that both servers use.   And same tag on
> both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
> > > had this working on opensips 2.4
> > >
> > > thanks
> > > Andy
> > >
> > >
> > > On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <
> kworm at missouri-telecom.com> wrote:
> > > Hi,
> > >
> > > I've been attempting to get a two node active/active setup to work
> with the v3.1 clusterer module sharing usrloc and dialog.  The setup is
> fronted by a proxy that handles all of the NAT/media so either OpenSIPS
> instance can communicate directly with the user.
> > >
> > > What I have working so far:
> > >
> > > Registrations and calls work when sent to either node and if you stop
> OpenSIPS on a node new calls work fine using the other node.
> > >
> > > What I can’t get to work:
> > >
> > > Calls that are already in progress to switch between nodes when one
> node fails.
> > >
> > >
> > > I have messed around with various sharing tags…no tag, same tag,
> different tags but haven’t had any luck.   I’m guessing that I’m missing
> something to trigger the remaining node to send re-invites.  Has anyone
> attempted this type of setup and have any ideas?
> > >
> > > Thanks,
> > >
> > > Kevin
> > > _______________________________________________
> > > 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
> > _______________________________________________
> > 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 social at bohboh.info  Tue Jan 19 15:31:37 2021
From: social at bohboh.info (Social Boh)
Date: Tue, 19 Jan 2021 10:31:37 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
Message-ID: <fa06b81c-37e5-aa3e-0651-96c08b0183c6@bohboh.info>

To switch calls from one server to another you have to use redis and 
rptengine using HA with pacemaker y corosync.

You must have two OpenSIPs, Two RTPEngine, Two Redis servers 
(primary-replica) Two Mariad  servers (primary/primary)

With redis you can save calls data (ip, ports, callid) on active server 
and then use these data on the replica server when swithc to active. On 
my tests, when switching from a server to another I have between 5 and 
10 seconds without audio.

Regards

---
I'm SoCIaL, MayBe

El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
> I’m not using a VIP and I have made some progress by setting a different active tag on each node…then upon node failure setting the failed node's tag to active on remaining node.  This lets the re-invite pinging work, etc.  It’s almost there but the handling of the BYE…they are still sent to the IP of the failed node even after re-invite pings so any in-progress calls from the failed node are zombie when they hang up until the re-invite ping times out (30 seconds).   I found an article about initiating a re-invite on the new node with something like "opensips-cli -x mi dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but that either seems to terminate the call immediately or say the dialog wasn’t found.
>
>
> Thanks,
>
> Kevin
>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
>>
>> With dialog writing to db that both servers use.   And same tag on both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
>> had this working on opensips 2.4
>>
>> thanks
>> Andy
>>
>>
>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com> wrote:
>> Hi,
>>
>> I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.
>>
>> What I have working so far:
>>
>> Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.
>>
>> What I can’t get to work:
>>
>> Calls that are already in progress to switch between nodes when one node fails.
>>
>>
>> I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?
>>
>> Thanks,
>>
>> Kevin
>> _______________________________________________
>> 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 liviu at opensips.org  Tue Jan 19 15:38:58 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 19 Jan 2021 17:38:58 +0200
Subject: [OpenSIPS-Users] Global variable $rm gives number when using
 $json
In-Reply-To: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>
References: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>
Message-ID: <866a6f7c-a8f2-d83c-e6bf-1c838d15f931@opensips.org>

On 18.01.2021 20:53, Alex Kinch wrote:
> I get this (identifying data redacted):
>
> Jan 18 18:18:15 [33] request: INVITE from sip:XX at XX to sip:XX at XX
> Jan 18 18:18:15 [33] Sending { "call_id": 
> "942887463-1604195939-951239259", "ts": "2021-01-18 18:18:15Z", 
> "src_ip": "XX", "dst_ip": "XX", "method": 1, "sip_from": "sip:XX at XX", 
> "sip_to": "sip:XX at XX", "dialled": "XX" }
>
> Any suggestions?
>
Hi Alex,

Thank you for the examples - indeed, that behavior is broken.

I just pushed a fix for this on "master" branch [1]. However, I'm a bit 
reluctant to backport it for the moment, because I haven't fully 
assessed its implications.  For example, could it be possible that 
people have already written code that *relie**s* on $json incorrectly 
returning the integer value of a variable which holds both a string and 
an integer, with string taking precedence (e.g. $rm)?

[1]: https://github.com/OpenSIPS/opensips/commit/6191f278a4

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com

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From kworm at missouri-telecom.com  Tue Jan 19 15:40:20 2021
From: kworm at missouri-telecom.com (Kevin Wormington)
Date: Tue, 19 Jan 2021 09:40:20 -0600
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <fa06b81c-37e5-aa3e-0651-96c08b0183c6@bohboh.info>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
 <fa06b81c-37e5-aa3e-0651-96c08b0183c6@bohboh.info>
Message-ID: <FAC36B73-76CF-479C-A6CE-F645C4679C95@missouri-telecom.com>

I’m not using RTPEngine…the upstream proxies are handling all media, NAT traversal, etc. so the OpenSIPS instances can always reach the endpoints.  I’m using clusterer module to share the user location and dialogs with different active tags per node.  There is zero loss of media on switch-over and sometimes a little longer PDD for new calls during switchover until the upstream proxies detect the instance down.   The only part I can’t seem to get to work is handling the final BYE for calls that were on the failed node originally.   The re-invite ping will correct end them but would like to be able to fix it completely…but maybe that is not currently possible.


Thanks,

Kevin
> On Jan 19, 2021, at 9:31 AM, Social Boh via Users <users at lists.opensips.org> wrote:
> 
> To switch calls from one server to another you have to use redis and rptengine using HA with pacemaker y corosync.
> 
> You must have two OpenSIPs, Two RTPEngine, Two Redis servers (primary-replica) Two Mariad  servers (primary/primary)
> 
> With redis you can save calls data (ip, ports, callid) on active server and then use these data on the replica server when swithc to active. On my tests, when switching from a server to another I have between 5 and 10 seconds without audio.
> 
> Regards
> 
> ---
> I'm SoCIaL, MayBe
> 
> El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
>> I’m not using a VIP and I have made some progress by setting a different active tag on each node…then upon node failure setting the failed node's tag to active on remaining node.  This lets the re-invite pinging work, etc.  It’s almost there but the handling of the BYE…they are still sent to the IP of the failed node even after re-invite pings so any in-progress calls from the failed node are zombie when they hang up until the re-invite ping times out (30 seconds).   I found an article about initiating a re-invite on the new node with something like "opensips-cli -x mi dlg_send_sequential callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge body=outbound” but that either seems to terminate the call immediately or say the dialog wasn’t found.
>> 
>> 
>> Thanks,
>> 
>> Kevin
>>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com> wrote:
>>> 
>>> With dialog writing to db that both servers use.   And same tag on both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
>>> had this working on opensips 2.4
>>> 
>>> thanks
>>> Andy
>>> 
>>> 
>>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington <kworm at missouri-telecom.com> wrote:
>>> Hi,
>>> 
>>> I've been attempting to get a two node active/active setup to work with the v3.1 clusterer module sharing usrloc and dialog.  The setup is fronted by a proxy that handles all of the NAT/media so either OpenSIPS instance can communicate directly with the user.
>>> 
>>> What I have working so far:
>>> 
>>> Registrations and calls work when sent to either node and if you stop OpenSIPS on a node new calls work fine using the other node.
>>> 
>>> What I can’t get to work:
>>> 
>>> Calls that are already in progress to switch between nodes when one node fails.
>>> 
>>> 
>>> I have messed around with various sharing tags…no tag, same tag, different tags but haven’t had any luck.   I’m guessing that I’m missing something to trigger the remaining node to send re-invites.  Has anyone attempted this type of setup and have any ideas?
>>> 
>>> Thanks,
>>> 
>>> Kevin
>>> _______________________________________________
>>> 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
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From social at bohboh.info  Tue Jan 19 16:00:21 2021
From: social at bohboh.info (Social Boh)
Date: Tue, 19 Jan 2021 11:00:21 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <FAC36B73-76CF-479C-A6CE-F645C4679C95@missouri-telecom.com>
References: <6CFC2A41-B6F0-42E2-9EF5-C64557040891@missouri-telecom.com>
 <CAKe=zzh5S_PrBA9LcXnwr5Jv5hWdWx7rpB+4X_03JzSjxd4ziA@mail.gmail.com>
 <3D389410-DCF7-4833-85E3-478F2ED04972@missouri-telecom.com>
 <fa06b81c-37e5-aa3e-0651-96c08b0183c6@bohboh.info>
 <FAC36B73-76CF-479C-A6CE-F645C4679C95@missouri-telecom.com>
Message-ID: <57f2a6a5-f51c-422e-d4db-ad6f3c234e72@bohboh.info>

I think your best option is KeepAlived; on keepalived configuration you 
declare a script name where you execute:

/usr/local/bin/opensips-cli -x mi clusterer_shtag_set_active vip/3

to switch VIP TAG from one server to other.

In this case BYE go to the right place.

If anyone want translate from spanish to english, I have a complete 
tutorial for OpenSIPs 3.1

Regards

---
I'm SoCIaL, MayBe

El 19/01/2021 a las 10:40 a. m., Kevin Wormington escribió:
> I’m not using RTPEngine…the upstream proxies are handling all media, 
> NAT traversal, etc. so the OpenSIPS instances can always reach the 
> endpoints.  I’m using clusterer module to share the user location and 
> dialogs with different active tags per node.  There is zero loss of 
> media on switch-over and sometimes a little longer PDD for new calls 
> during switchover until the upstream proxies detect the instance down. 
>   The only part I can’t seem to get to work is handling the final BYE 
> for calls that were on the failed node originally.   The re-invite 
> ping will correct end them but would like to be able to fix it 
> completely…but maybe that is not currently possible.
>
>
> Thanks,
>
> Kevin
>> On Jan 19, 2021, at 9:31 AM, Social Boh via Users 
>> <users at lists.opensips.org <mailto:users at lists.opensips.org>> wrote:
>>
>> To switch calls from one server to another you have to use redis and 
>> rptengine using HA with pacemaker y corosync.
>>
>> You must have two OpenSIPs, Two RTPEngine, Two Redis servers 
>> (primary-replica) Two Mariad  servers (primary/primary)
>>
>> With redis you can save calls data (ip, ports, callid) on active 
>> server and then use these data on the replica server when swithc to 
>> active. On my tests, when switching from a server to another I have 
>> between 5 and 10 seconds without audio.
>>
>> Regards
>>
>> ---
>> I'm SoCIaL, MayBe
>>
>> El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
>>> I’m not using a VIP and I have made some progress by setting a 
>>> different active tag on each node…then upon node failure setting the 
>>> failed node's tag to active on remaining node.  This lets the 
>>> re-invite pinging work, etc.  It’s almost there but the handling of 
>>> the BYE…they are still sent to the IP of the failed node even after 
>>> re-invite pings so any in-progress calls from the failed node are 
>>> zombie when they hang up until the re-invite ping times out (30 
>>> seconds).   I found an article about initiating a re-invite on the 
>>> new node with something like "opensips-cli -x mi dlg_send_sequential 
>>> callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge 
>>> body=outbound” but that either seems to terminate the call 
>>> immediately or say the dialog wasn’t found.
>>>
>>>
>>> Thanks,
>>>
>>> Kevin
>>>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com 
>>>> <mailto:adierlam at ptgi-ics.com>> wrote:
>>>>
>>>> With dialog writing to db that both servers use.   And same tag on 
>>>> both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
>>>> had this working on opensips 2.4
>>>>
>>>> thanks
>>>> Andy
>>>>
>>>>
>>>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington 
>>>> <kworm at missouri-telecom.com <mailto:kworm at missouri-telecom.com>> wrote:
>>>> Hi,
>>>>
>>>> I've been attempting to get a two node active/active setup to work 
>>>> with the v3.1 clusterer module sharing usrloc and dialog.  The 
>>>> setup is fronted by a proxy that handles all of the NAT/media so 
>>>> either OpenSIPS instance can communicate directly with the user.
>>>>
>>>> What I have working so far:
>>>>
>>>> Registrations and calls work when sent to either node and if you 
>>>> stop OpenSIPS on a node new calls work fine using the other node.
>>>>
>>>> What I can’t get to work:
>>>>
>>>> Calls that are already in progress to switch between nodes when one 
>>>> node fails.
>>>>
>>>>
>>>> I have messed around with various sharing tags…no tag, same tag, 
>>>> different tags but haven’t had any luck.   I’m guessing that I’m 
>>>> missing something to trigger the remaining node to send re-invites. 
>>>>  Has anyone attempted this type of setup and have any ideas?
>>>>
>>>> Thanks,
>>>>
>>>> Kevin
>>>> _______________________________________________
>>>> 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
>>>
>>> _______________________________________________
>>> 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 kevin.vines at gmail.com  Tue Jan 19 17:02:22 2021
From: kevin.vines at gmail.com (Kevin Vines)
Date: Tue, 19 Jan 2021 12:02:22 -0500
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <57f2a6a5-f51c-422e-d4db-ad6f3c234e72@bohboh.info>
Message-ID: <orpf0dh32u5n0m49ct01qqth.1611075742840@gmail.com>

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From johan at democon.be  Tue Jan 19 17:10:56 2021
From: johan at democon.be (Johan De Clercq)
Date: Tue, 19 Jan 2021 17:10:56 +0000
Subject: [OpenSIPS-Users] Global variable $rm gives number when using
 $json
In-Reply-To: <866a6f7c-a8f2-d83c-e6bf-1c838d15f931@opensips.org>
References: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>,
 <866a6f7c-a8f2-d83c-e6bf-1c838d15f931@opensips.org>
Message-ID: <PR1PR06MB468304A8B8D4AAD84F842058A7A30@PR1PR06MB4683.eurprd06.prod.outlook.com>

Liviu,

For that case, why don’t you make an extra parameter ? E.g. use-broken-implementation 0|1

Outlook voor iOS<https://aka.ms/o0ukef> downloaden
________________________________
Van: Users <users-bounces at lists.opensips.org> namens Liviu Chircu <liviu at opensips.org>
Verzonden: Tuesday, January 19, 2021 4:38:58 PM
Aan: OpenSIPS users mailling list <users at lists.opensips.org>
Onderwerp: Re: [OpenSIPS-Users] Global variable $rm gives number when using $json

On 18.01.2021 20:53, Alex Kinch wrote:
I get this (identifying data redacted):

Jan 18 18:18:15 [33] request: INVITE from sip:XX at XX to sip:XX at XX
Jan 18 18:18:15 [33] Sending { "call_id": "942887463-1604195939-951239259", "ts": "2021-01-18 18:18:15Z", "src_ip": "XX", "dst_ip": "XX", "method": 1, "sip_from": "sip:XX at XX", "sip_to": "sip:XX at XX", "dialled": "XX" }

Any suggestions?


Hi Alex,

Thank you for the examples - indeed, that behavior is broken.

I just pushed a fix for this on "master" branch [1].  However, I'm a bit reluctant to backport it for the moment, because I haven't fully assessed its implications.  For example, could it be possible that people have already written code that relies on $json incorrectly returning the integer value of a variable which holds both a string and an integer, with string taking precedence (e.g. $rm)?

[1]: https://github.com/OpenSIPS/opensips/commit/6191f278a4

--
Liviu Chircu
www.twitter.com/liviuchircu<http://www.twitter.com/liviuchircu> | www.opensips-solutions.com<http://www.opensips-solutions.com>
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From liviu at opensips.org  Tue Jan 19 17:17:16 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 19 Jan 2021 19:17:16 +0200
Subject: [OpenSIPS-Users] Global variable $rm gives number when using
 $json
In-Reply-To: <PR1PR06MB468304A8B8D4AAD84F842058A7A30@PR1PR06MB4683.eurprd06.prod.outlook.com>
References: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>
 <866a6f7c-a8f2-d83c-e6bf-1c838d15f931@opensips.org>
 <PR1PR06MB468304A8B8D4AAD84F842058A7A30@PR1PR06MB4683.eurprd06.prod.outlook.com>
Message-ID: <3e3d4025-7d8c-2206-8e9d-d91fc4e93d0e@opensips.org>

On 19.01.2021 19:10, Johan De Clercq wrote:
> Liviu,
>
> For that case, why don’t you make an extra parameter ? E.g. 
> use-broken-implementation 0|1
>
Sounds good for 2.4, which is heavily used out there and the risks are 
bigger, thanks!

For 3.1, I think we're still in the right time window to backport this 
fix without breaking any opensips.cfg files.

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com



From alex at alexkinch.com  Tue Jan 19 17:22:42 2021
From: alex at alexkinch.com (Alex Kinch)
Date: Tue, 19 Jan 2021 17:22:42 +0000
Subject: [OpenSIPS-Users] Global variable $rm gives number when using
	$json
In-Reply-To: <3e3d4025-7d8c-2206-8e9d-d91fc4e93d0e@opensips.org>
References: <CAByD3_tgV8cAEmX7LNEPqFEScm1vKE9NvwQy0wXD2xfyUUd7NQ@mail.gmail.com>
 <866a6f7c-a8f2-d83c-e6bf-1c838d15f931@opensips.org>
 <PR1PR06MB468304A8B8D4AAD84F842058A7A30@PR1PR06MB4683.eurprd06.prod.outlook.com>
 <3e3d4025-7d8c-2206-8e9d-d91fc4e93d0e@opensips.org>
Message-ID: <CAByD3_sZD+1jM2sNkuwJ4Tw8Q5n6uu+ipdMROSaCUVxepgAgvQ@mail.gmail.com>

On Tue, 19 Jan 2021 at 17:17, Liviu Chircu <liviu at opensips.org> wrote:


> For 3.1, I think we're still in the right time window to backport this
> fix without breaking any opensips.cfg files.


I'm using 3.1 so quite happy with that proposal :) Thanks for the fast
response and fix, much appreciated.

Alex
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From donat.zenichev at gmail.com  Wed Jan 20 07:14:56 2021
From: donat.zenichev at gmail.com (Donat Zenichev)
Date: Wed, 20 Jan 2021 09:14:56 +0200
Subject: [OpenSIPS-Users] www_challenge qop type failed with AAA
In-Reply-To: <0235E4F6-C75B-4C96-A46D-4159A52CD7F2@vale.ski>
References: <E03414AB-518C-4CF7-93E4-0923EFB587BC@vale.ski>
 <2dc8c2b8-1df1-2cda-9d6e-51ab65bb86f6@opensips.org>
 <0235E4F6-C75B-4C96-A46D-4159A52CD7F2@vale.ski>
Message-ID: <CANLwQCn8WaQ+xP3rqvpq88pibGHw=BssoakOVUEth6HGxcqGvQ@mail.gmail.com>

Good day Michael,
I just noticed you are using quotation marks when you pass the second
parameter to the function www_challenge().

Reading the manual on auth.so:
https://opensips.org/html/docs/modules/3.0.x/auth.html#func_www_challenge

I see that it should be passed without any preceding and following symbols,
so clear 0 or 1, see example from module's wiki:
www_challenge("siphub.net", 1);

On Tue, Jan 5, 2021 at 8:35 PM Michael Vale via Users <
users at lists.opensips.org> wrote:

> The script wouldn’t load if I specified a auth-int as well as auth.  I
> removed auth-int entirely and the script runs.
>
> > On 6 Jan 2021, at 5:26 am, Vlad Patrascu <vladp at opensips.org> wrote:
> >
> > Hi Michael,
> >
> > Regarding the qop issue, you should pass for that parameter a string
> value of "auth", "auth-int", or both separated by ','.
> >
> > Regards,
> >
> > --
> > Vlad Patrascu
> > OpenSIPS Developer
> > http://www.opensips-solutions.com
> >
> > On 05.01.2021 19:50, Michael Vale via Users wrote:
> >> Hi,
> >>
> >> I’m attempting to do auth with AAA,
> >>
> >> I was met with an error of no Cisco vendor and no SIP-URI-Host
> Attribute which I added to the dictionaries.
> >>
> >> Now www_challenge thinks either “0” or “1” is an invalid qop type.
>
> >>
> >>
>        ERROR:auth:fixup_qop: Bad qop type
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:fix_cmd:
> Fixup failed for param [2]
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]:
> ERROR:core:fix_actions: Failed to fix command <www_challenge>
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]:
> ERROR:core:fix_actions: fixing failed (code=-1) at
> /etc/opensips/opensips.cfg:216
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: ERROR:core:main:
> failed to fix configuration with err code -1
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: INFO:core:cleanup:
> cleanup
> >> Jan 06 04:42:14 unispy /usr/sbin/opensips[29711]: NOTICE:core:main:
> Exiting....
> >> Jan 06 04:42:14 unispy opensips[29709]: INFO:core:daemonize: pre-daemon
> proce
> >>
> >>      if (!aaa_www_authorize("")) {
> >>             www_challenge("", "1");
> >>          };
> >>
> >>
> >> Could someone please shed some light on the subject?
> >>
> >> Also it would be nice if there was another approach to handling the
> SIP-URI-Host and Cisco issue or at least some confirmation that my approach
> was the best way to deal with it.
> >>
> >>
> >> Regards,
> >>
> >> Michael.
> >> _______________________________________________
> >> 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
>


-- 

Best regards,
Donat Zenichev
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From donat.zenichev at gmail.com  Wed Jan 20 08:03:10 2021
From: donat.zenichev at gmail.com (Donat Zenichev)
Date: Wed, 20 Jan 2021 10:03:10 +0200
Subject: [OpenSIPS-Users] Mediaproxy configuration
In-Reply-To: <CADaqb1u6stAKbTaQJ4ru=2txyTOjWqhfEvEfWOn-QrMLtrC3Vw@mail.gmail.com>
References: <CADaqb1s4Aznb6Wmp2Vpq+QbMT6wniMZJOdu-QNk3X=THofN9CQ@mail.gmail.com>
 <CADaqb1u6stAKbTaQJ4ru=2txyTOjWqhfEvEfWOn-QrMLtrC3Vw@mail.gmail.com>
Message-ID: <CANLwQCmGH2YjDsZT1OCTAtuW0JPEwC=nk3C0nqQqmrf86xDUYA@mail.gmail.com>

Good day Mark,
you have an interesting case actually.

How does the mapping at (IPA-IPB setup) work, when traffic goes back from
Media proxy to your user agent?
Let's imagine that the media proxy starts sending RTP/RTCP from
192.168.xxx.xxx,
then it reaches the map server, and which ports are then allocated for the
public side? Is that the same as the Media proxy allocated? (and advertised
in SDP as well)
And what if these ports for UDP transport are already engaged, how IPA-IPB
setup then manages this?

On the other hand, let's imagine your user agent sends an SDP offer in the
initial request.
Even though it advertised not a private address and there is no NAT problem
at UAC's side,
the contact information given in the SDP body will be the address which
should be reachable for your Media proxy server,
since this is what your Media proxy sees when receiving the offer.
(if I understand your description properly, then there is no entity which
would fix SDP body coming from IPA-IPB setup to Media Proxy)

If Media proxy received the local address of your test user agent (which is
even a public address), then it should have a possibility to reach it over
the IP network.
How does the RTP/RTCP flow go in this case? (from Media proxy of course)

Another good question, did you take a look at SDP bodies of both user agent
and Media proxy?
It's always a good thing to investigate media attributes, and other basic
information.




On Thu, Jan 7, 2021 at 2:57 PM Mark Allen <mark at allenclan.co.uk> wrote:

> Sorry... should have added that OpenSIPS box is acting as mid-registrar
>
> On Thu, 7 Jan 2021, 12:12 Mark Allen, <mark at allenclan.co.uk> wrote:
>
>> I wonder if anyone can help me with this? I am trying to configure
>> Mediaproxy to handle RTP traffic coming from outside our local network.
>> Here's the setup:
>>
>>     UAC ---> IPA ---> IPB ---> Mediaproxy / OpenSIPS ---> Asterisk
>>
>> IPA (a public IP address 4x.xxx.xxx.xxx) maps ports ports 5060 and 10000
>> to 65535 to IPB (local IP address 192.168.xxx.xxx). IPB is actually a
>> Virtual IP managed by keepalived.
>> UAC is MizuDroid app running on my Android phone connected to my home
>> network (NATed) with a public IP of 5.xxx.xxx.xxx. Everything else relates
>> to our office network.
>> Mediaproxy Dispatcher and Relay are both running on the same (OpenSIPS)
>> system
>>
>> SIP conversation between UAC and Asterisk via OpenSIPS looks to be
>> working fine. Endpoints connect, exchange data, and hangup. The problem is
>> with SDP addressing (NAT problem) causing no audio either way, which is
>> what I want Mediaproxy to handle.
>>
>> In opensips.cfg I'm passing control for calls arriving at IPA to
>> Mediaproxy...
>>
>>     if (is_method("INVITE")) {
>>         if (!has_totag()) {
>>             if ($fd == "4x.xxx.xxx.xxx") {
>>                 xlog("Passing control to Mediaproxy...");
>>                 engage_media_proxy();
>>             }
>>         }
>>     }
>>
>> In /etc/mediaproxy/config.ini all settings are defaults except for
>> setting dispatcher as IPB...
>>
>>     dispatchers = 192.168.xxx.xxx
>>
>> ...and I've tried it with and without advertised_ip set to IPA...
>>
>>     advertised_ip = 4x.xxx.xxx.xxx
>>
>>
>> I can see that Mediaproxy is taking control of calls as instructed and
>> making changes to SDP but it's not solving my audio problems. What am I
>> doing wrong????
>>
>>
>>
>>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 

Best regards,
Donat Zenichev
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From alexanderhenryperkins at gmail.com  Thu Jan 21 03:00:45 2021
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Wed, 20 Jan 2021 22:00:45 -0500
Subject: [OpenSIPS-Users] Limit Call per Second
Message-ID: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>

Hi All.  Is there a way to limit the calls per second by pulling the
information from a database? I was looking at the call_control module and
that seems to be a global value. but I would need something that I can
control at the call level (I will be handling different accounts that have
different limits).

Thanks, All.  Any help is appreciated.

Alex
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From jehanzaib.kiani at gmail.com  Thu Jan 21 03:07:28 2021
From: jehanzaib.kiani at gmail.com (Jehanzaib Younis)
Date: Thu, 21 Jan 2021 16:07:28 +1300
Subject: [OpenSIPS-Users] Limit Call per Second
In-Reply-To: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
References: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
Message-ID: <CAKrJzCzMq9JEKCY7QxjuFAHgA=Uw=e1jMLiY4sADEs3AcJMz+w@mail.gmail.com>

Hi Alex,

You can save the data from your database to memory and fetch it during all
control. I will not recommend to use db directly. There might be any other
better way someone from the community can tell us ;)


Regards,
Jehanzaib


On Thu, Jan 21, 2021 at 4:01 PM Alexander Perkins <
alexanderhenryperkins at gmail.com> wrote:

> Hi All.  Is there a way to limit the calls per second by pulling the
> information from a database? I was looking at the call_control module and
> that seems to be a global value. but I would need something that I can
> control at the call level (I will be handling different accounts that have
> different limits).
>
> Thanks, All.  Any help is appreciated.
>
> Alex
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From osas at voipembedded.com  Thu Jan 21 03:41:31 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 20 Jan 2021 22:41:31 -0500
Subject: [OpenSIPS-Users] Limit Call per Second
In-Reply-To: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
References: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
Message-ID: <CAND0LksvhjUzTxh9F9Q-rMF8sB_exD_HSJ70x3mw+0t=wWc7_g@mail.gmail.com>

Take a look at the ratelimit module:
https://opensips.org/docs/modules/3.1.x/ratelimit.html

-ovidiu

On Wed, Jan 20, 2021 at 10:01 PM Alexander Perkins
<alexanderhenryperkins at gmail.com> wrote:
>
> Hi All.  Is there a way to limit the calls per second by pulling the information from a database? I was looking at the call_control module and that seems to be a global value. but I would need something that I can control at the call level (I will be handling different accounts that have different limits).
>
> Thanks, All.  Any help is appreciated.
>
> Alex
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com


From john.quick at smartvox.co.uk  Thu Jan 21 09:40:18 2021
From: john.quick at smartvox.co.uk (John Quick)
Date: Thu, 21 Jan 2021 09:40:18 -0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
Message-ID: <001c01d6efd9$71d7fb50$5587f1f0$@smartvox.co.uk>

Mark,

I recommend using rtpproxy (or possibly rtpengine) rather than mediaproxy
for your situation.
You need the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server.
This is what bridge mode (or bridging mode) is used for, although the last
time I built a solution like this I didn't use bridge mode and instead
passed the relevant IP address as an argument when calling the rtpproxy
activation functions. Unfortunately, the latter approach means your
opensips.cfg script will need to be much more complicated.

I suspect your problem when using mediaproxy and advertised_ip =
4x.xxx.xxx.xxx is that it will pass that address to Asterisk in the SDP. In
which case, you might be able to get audio if you look at the network route
Asterisk would use to reach/connect to 4x.xxx.xxx.xxx and make sure the
mediaproxy relay is reachable. However, that does not sound like a good
solution to me - much better if Asterisk talks to the relay directly over
the LAN.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk




From johan at democon.be  Thu Jan 21 10:21:05 2021
From: johan at democon.be (Johan De Clercq)
Date: Thu, 21 Jan 2021 10:21:05 +0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
In-Reply-To: <001c01d6efd9$71d7fb50$5587f1f0$@smartvox.co.uk>
References: <001c01d6efd9$71d7fb50$5587f1f0$@smartvox.co.uk>
Message-ID: <PR1PR06MB4683C68851C095F4CB03C9D8A7A10@PR1PR06MB4683.eurprd06.prod.outlook.com>

I totally agree with the rtpengine suggestion

Outlook voor iOS<https://aka.ms/o0ukef> downloaden
________________________________
Van: Users <users-bounces at lists.opensips.org> namens John Quick <john.quick at smartvox.co.uk>
Verzonden: Thursday, January 21, 2021 10:40:18 AM
Aan: users at lists.opensips.org <users at lists.opensips.org>
Onderwerp: Re: [OpenSIPS-Users] Mediaproxy configuration

Mark,

I recommend using rtpproxy (or possibly rtpengine) rather than mediaproxy
for your situation.
You need the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server.
This is what bridge mode (or bridging mode) is used for, although the last
time I built a solution like this I didn't use bridge mode and instead
passed the relevant IP address as an argument when calling the rtpproxy
activation functions. Unfortunately, the latter approach means your
opensips.cfg script will need to be much more complicated.

I suspect your problem when using mediaproxy and advertised_ip =
4x.xxx.xxx.xxx is that it will pass that address to Asterisk in the SDP. In
which case, you might be able to get audio if you look at the network route
Asterisk would use to reach/connect to 4x.xxx.xxx.xxx and make sure the
mediaproxy relay is reachable. However, that does not sound like a good
solution to me - much better if Asterisk talks to the relay directly over
the LAN.

John Quick
Smartvox Limited
Web: www.smartvox.co.uk<http://www.smartvox.co.uk>



_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From farmorg at gmail.com  Thu Jan 21 12:27:25 2021
From: farmorg at gmail.com (Mark Farmer)
Date: Thu, 21 Jan 2021 12:27:25 +0000
Subject: [OpenSIPS-Users] How to see outgoing TLS OPTIONS
In-Reply-To: <CAOvQDet88BTTUk++F432Ra2deVvv76LRa=u-zjqgL-bQ+MenAg@mail.gmail.com>
References: <CAOvQDet+d9w1Vq0OAe3WWM0f06qgiTbsSbErWWO-P+oUFU7KTg@mail.gmail.com>
 <CAOvQDet88BTTUk++F432Ra2deVvv76LRa=u-zjqgL-bQ+MenAg@mail.gmail.com>
Message-ID: <CAOvQDes9zFffrK-M+Dac+juGD3Mnu1mqDGrJKg_D23PzNmsWqg@mail.gmail.com>

Hi everyone

In a recent thread I learnt that using trace() in local_route after
appending a Contact header traces the message before the new header is
actually added which means I don't get to see it.

Is there a way to see the OPTIONS message after the new header is added?

Many thanks
Mark.


On Thu, 12 Mar 2020 at 13:02, Mark Farmer <farmorg at gmail.com> wrote:

> I think I answered my own question :)
>
> Adding a sip_trace() into local_route seems to do the trick :)
>
> Mark.
>
>
> On Thu, 12 Mar 2020 at 12:31, Mark Farmer <farmorg at gmail.com> wrote:
>
>> Hi everyone
>>
>> I am using the drouting module to make SIP/TLS connections and I need to
>> be able to capture the outgoing OPTIONS requests generated by drouting.
>>
>> I am thinking sip_trace("hep_dst", "d"); but where would I need to do
>> that?
>>
>> Is there a better way?
>>
>> OpenSIPS 2.4
>>
>> Many thanks!
>> Mark.
>>
>>
>
> --
> Mark Farmer
> farmorg at gmail.com
>


-- 
Mark Farmer
farmorg at gmail.com
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From Johan at democon.be  Thu Jan 21 14:55:20 2021
From: Johan at democon.be (Johan De Clercq)
Date: Thu, 21 Jan 2021 15:55:20 +0100
Subject: [OpenSIPS-Users] How to see outgoing TLS OPTIONS
In-Reply-To: <CAOvQDes9zFffrK-M+Dac+juGD3Mnu1mqDGrJKg_D23PzNmsWqg@mail.gmail.com>
References: <CAOvQDet+d9w1Vq0OAe3WWM0f06qgiTbsSbErWWO-P+oUFU7KTg@mail.gmail.com>
 <CAOvQDet88BTTUk++F432Ra2deVvv76LRa=u-zjqgL-bQ+MenAg@mail.gmail.com>
 <CAOvQDes9zFffrK-M+Dac+juGD3Mnu1mqDGrJKg_D23PzNmsWqg@mail.gmail.com>
Message-ID: <CAEVWGm_4kwNS2SEsV1a+EWs3fMhRuU1wTQv9-5aGZpgxqtxgJw@mail.gmail.com>

Install Homer with portmirror on switch

On Thu, Jan 21, 2021, 13:31 Mark Farmer <farmorg at gmail.com> wrote:

> Hi everyone
>
> In a recent thread I learnt that using trace() in local_route after
> appending a Contact header traces the message before the new header is
> actually added which means I don't get to see it.
>
> Is there a way to see the OPTIONS message after the new header is added?
>
> Many thanks
> Mark.
>
>
> On Thu, 12 Mar 2020 at 13:02, Mark Farmer <farmorg at gmail.com> wrote:
>
>> I think I answered my own question :)
>>
>> Adding a sip_trace() into local_route seems to do the trick :)
>>
>> Mark.
>>
>>
>> On Thu, 12 Mar 2020 at 12:31, Mark Farmer <farmorg at gmail.com> wrote:
>>
>>> Hi everyone
>>>
>>> I am using the drouting module to make SIP/TLS connections and I need to
>>> be able to capture the outgoing OPTIONS requests generated by drouting.
>>>
>>> I am thinking sip_trace("hep_dst", "d"); but where would I need to do
>>> that?
>>>
>>> Is there a better way?
>>>
>>> OpenSIPS 2.4
>>>
>>> Many thanks!
>>> Mark.
>>>
>>>
>>
>> --
>> Mark Farmer
>> farmorg at gmail.com
>>
>
>
> --
> Mark Farmer
> farmorg at gmail.com
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From jofi.yance at gmail.com  Thu Jan 21 16:24:42 2021
From: jofi.yance at gmail.com (jofi Y)
Date: Thu, 21 Jan 2021 17:24:42 +0100
Subject: [OpenSIPS-Users] Siprec contact header
Message-ID: <CAMo3uQwM8-B6w-wSzB1q1oZ0tSt_qjDr54emkEREv9fsB3Wd3Q@mail.gmail.com>

Hello All,

I have a question regarding contact header generated by siprec module.
According to rfc7866 (6.1.1):

  The SRC MUST include the "+sip.src" feature tag in the Contact URI,
   defined in this specification as an extension to [RFC3840
<https://tools.ietf.org/html/rfc3840>], for all
   RSs.


The contact parameter "+sip.src" is not generated when I started recording.
Currently I use Opensips version 2.4.6 and 3.1.1.

Is there an alternative way to generate "+sip.src" when recording session
started?

Many thanks,
Jofi
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From chunyong.zhao at qq.com  Fri Jan 22 02:57:26 2021
From: chunyong.zhao at qq.com (=?ISO-8859-1?B?SmVmZnJleSBaaGFv?=)
Date: Fri, 22 Jan 2021 10:57:26 +0800
Subject: [OpenSIPS-Users] Question regarding Federated User Location Cluster
Message-ID: <tencent_2BDF660DD375931304A79016DD3F8B7DDE06@qq.com>

Dear Team
After reading through below Tutorial, I have some questions regarding Database and NoSQL setup model for Federated User Location Cluster.




https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation


For example, for formal production system senario, two sites, with two opensips nodes for each site, HA mode for each site.
1. Should I deploy MySQL and Cassandra on each node? 4 MySQL instances and 4 Cassandra on each node?
2. Should I setup db replication among 4 MySQL instances or just standalone separated setup?
3. Should I put all 4&nbsp; Cassandra instances into one cluster?
4. For each HA virtual IP setup, what's the recommended tool, keepalived?




Thanks in advance.


Best wishes,
Jeffrey
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From spanda at 3clogic.com  Fri Jan 22 08:14:23 2021
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 22 Jan 2021 13:44:23 +0530
Subject: [OpenSIPS-Users] Need some help for auth_jwt module .
Message-ID: <CALf-Q46n528Cz40TXH7uSjcDGe+VJJeZ4JauwjZFjNF9GoTLYw@mail.gmail.com>

Hi ,
First time I am trying to use auth_jwt module with opensips 3.1 .

When I am generating jwt token by using HS256 algorithm and adding the
corresponding tag and key in opensips DB then opensips is able to decode
the token successfully .

When I am generating the token by using RS256 algorithm  and both private
and public key in the corresponding to the token in the db then opensips is
not able to decode the token using that key .

 DBG:auth_jwt:jwt_authorize: Found 2 record for tag 1234567890
 DBG:auth_jwt:jwt_authorize: Failed to decode jwt with DB secret
 DBG:auth_jwt:jwt_authorize: Failed to decode jwt with DB secret

for both the secrets this is the error I am getting . How will I do this ?
What should I add in secret column corresponding to the token ?


*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From johan at democon.be  Fri Jan 22 08:15:46 2021
From: johan at democon.be (johan)
Date: Fri, 22 Jan 2021 09:15:46 +0100
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
Message-ID: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>

Hi,


when you use the siptrace module and you go to homer from opensips
(hence without a port mirror/span port on a switch), when you change
from /to header, then the homer trace doesnot reflect the real packet
that exits opensips (as we all know, from to header changes are only
active when the packet is send out). 


I find this annoying as the purpose of having a tracer is defeated by
this : we want to see the real packets with the changes applied.


Therefore : are there more people on the list who are ennoyed by this ?
If yes, then I will open a feature request for it.


wkr,

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From gmaruzz at gmail.com  Fri Jan 22 09:49:24 2021
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 22 Jan 2021 10:49:24 +0100
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
Message-ID: <CALXCt0oLq0-5i7V3g9V29Opd204Z9hQ3BtawfJoxhEPX4shuMg@mail.gmail.com>

+1


On Fri, Jan 22, 2021 at 9:18 AM johan <johan at democon.be> wrote:

> Hi,
>
>
> when you use the siptrace module and you go to homer from opensips
> (hence without a port mirror/span port on a switch), when you change
> from /to header, then the homer trace doesnot reflect the real packet
> that exits opensips (as we all know, from to header changes are only
> active when the packet is send out).
>
>
> I find this annoying as the purpose of having a tracer is defeated by
> this : we want to see the real packets with the changes applied.
>
>
> Therefore : are there more people on the list who are ennoyed by this ?
> If yes, then I will open a feature request for it.
>
>
> wkr,
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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From nick at altmann.pro  Fri Jan 22 11:13:18 2021
From: nick at altmann.pro (Nick Altmann)
Date: Fri, 22 Jan 2021 12:13:18 +0100
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
Message-ID: <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>

It depends on purpose. For example, I need an initial (untouched) packet
there.

--
Nick

пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be>:

> Hi,
>
>
> when you use the siptrace module and you go to homer from opensips
> (hence without a port mirror/span port on a switch), when you change
> from /to header, then the homer trace doesnot reflect the real packet
> that exits opensips (as we all know, from to header changes are only
> active when the packet is send out).
>
>
> I find this annoying as the purpose of having a tracer is defeated by
> this : we want to see the real packets with the changes applied.
>
>
> Therefore : are there more people on the list who are ennoyed by this ?
> If yes, then I will open a feature request for it.
>
>
> wkr,
>
> _______________________________________________
> 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  Fri Jan 22 11:34:28 2021
From: johan at democon.be (johan)
Date: Fri, 22 Jan 2021 12:34:28 +0100
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
Message-ID: <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>

yeah me too.

but I want also to see the forwarded packet with all changes.

On 22/01/2021 12:13, Nick Altmann wrote:
> It depends on purpose. For example, I need an initial (untouched)
> packet there.
>
> --
> Nick
>
> пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be
> <mailto:johan at democon.be>>:
>
>     Hi,
>
>
>     when you use the siptrace module and you go to homer from opensips
>     (hence without a port mirror/span port on a switch), when you change
>     from /to header, then the homer trace doesnot reflect the real packet
>     that exits opensips (as we all know, from to header changes are only
>     active when the packet is send out). 
>
>
>     I find this annoying as the purpose of having a tracer is defeated by
>     this : we want to see the real packets with the changes applied.
>
>
>     Therefore : are there more people on the list who are ennoyed by
>     this ?
>     If yes, then I will open a feature request for it.
>
>
>     wkr,
>
>     _______________________________________________
>     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 adierlam at ptgi-ics.com  Fri Jan 22 14:11:03 2021
From: adierlam at ptgi-ics.com (Andy Dierlam)
Date: Fri, 22 Jan 2021 09:11:03 -0500
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
 <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>
Message-ID: <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>

How about running a continuous tshark/pcap/ngrep on interfaces of interest.
you can run every 10min.  putting in 10min files.

On Fri, Jan 22, 2021 at 6:36 AM johan <johan at democon.be> wrote:

> yeah me too.
>
> but I want also to see the forwarded packet with all changes.
> On 22/01/2021 12:13, Nick Altmann wrote:
>
> It depends on purpose. For example, I need an initial (untouched) packet
> there.
>
> --
> Nick
>
> пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be>:
>
>> Hi,
>>
>>
>> when you use the siptrace module and you go to homer from opensips
>> (hence without a port mirror/span port on a switch), when you change
>> from /to header, then the homer trace doesnot reflect the real packet
>> that exits opensips (as we all know, from to header changes are only
>> active when the packet is send out).
>>
>>
>> I find this annoying as the purpose of having a tracer is defeated by
>> this : we want to see the real packets with the changes applied.
>>
>>
>> Therefore : are there more people on the list who are ennoyed by this ?
>> If yes, then I will open a feature request for it.
>>
>>
>> wkr,
>>
>> _______________________________________________
>> 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
>
> _______________________________________________
> 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 genesys.com  Fri Jan 22 14:28:39 2021
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Fri, 22 Jan 2021 14:28:39 +0000
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
 <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>,
 <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>
Message-ID: <BN7PR10MB259408FAC86EBF02AA21267C95A09@BN7PR10MB2594.namprd10.prod.outlook.com>

We are using siptrace and we get both the incoming and outbound message captured, with all changes. I think it depends on the scope you provide [1]. We use scope T for transactions and we get both.

Although, we are using OpenSIPS 2.4. It’s possible the functionality has changed in 3.x. In which case, I would agree that it is not good.

[1] - https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Andy Dierlam <adierlam at ptgi-ics.com>
Date: Friday, January 22, 2021 at 9:13 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] feasablity / interest in a feature request.
How about running a continuous tshark/pcap/ngrep on interfaces of interest.
you can run every 10min.  putting in 10min files.

On Fri, Jan 22, 2021 at 6:36 AM johan <johan at democon.be<mailto:johan at democon.be>> wrote:

yeah me too.

but I want also to see the forwarded packet with all changes.
On 22/01/2021 12:13, Nick Altmann wrote:
It depends on purpose. For example, I need an initial (untouched) packet there.

--
Nick

пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be<mailto:johan at democon.be>>:
Hi,


when you use the siptrace module and you go to homer from opensips
(hence without a port mirror/span port on a switch), when you change
from /to header, then the homer trace doesnot reflect the real packet
that exits opensips (as we all know, from to header changes are only
active when the packet is send out).


I find this annoying as the purpose of having a tracer is defeated by
this : we want to see the real packets with the changes applied.


Therefore : are there more people on the list who are ennoyed by this ?
If yes, then I will open a feature request for it.


wkr,

_______________________________________________
Users mailing list
Users at lists.opensips.org<mailto: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<mailto: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<mailto:Users at lists.opensips.org>
http://lists.opensips.org/cgi-bin/mailman/listinfo/users<http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
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From johan at democon.be  Fri Jan 22 14:36:46 2021
From: johan at democon.be (johan)
Date: Fri, 22 Jan 2021 15:36:46 +0100
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <BN7PR10MB259408FAC86EBF02AA21267C95A09@BN7PR10MB2594.namprd10.prod.outlook.com>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
 <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>
 <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>
 <BN7PR10MB259408FAC86EBF02AA21267C95A09@BN7PR10MB2594.namprd10.prod.outlook.com>
Message-ID: <bb76e3b3-fa21-a71e-0cf9-65f63581f6fc@democon.be>

I use this for invite based dialogs.

trace("hep_id","d","sip");

I call it just before topology_hiding()

I don't see from/to header changes.

Can you check on your side Ben ?


wkr,

On 22/01/2021 15:28, Ben Newlin wrote:
>
> We are using siptrace and we get both the incoming and outbound
> message captured, with all changes. I think it depends on the scope
> you provide [1]. We use scope T for transactions and we get both.
>
>  
>
> Although, we are using OpenSIPS 2.4. It’s possible the functionality
> has changed in 3.x. In which case, I would agree that it is not good.
>
>  
>
> [1] - https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace
>
>  
>
> Ben Newlin
>
>  
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Andy
> Dierlam <adierlam at ptgi-ics.com>
> *Date: *Friday, January 22, 2021 at 9:13 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] feasablity / interest in a feature
> request.
>
> How about running a continuous tshark/pcap/ngrep on interfaces of
> interest.
>
> you can run every 10min.  putting in 10min files.
>
>  
>
> On Fri, Jan 22, 2021 at 6:36 AM johan <johan at democon.be
> <mailto:johan at democon.be>> wrote:
>
>     yeah me too.
>
>     but I want also to see the forwarded packet with all changes.
>
>     On 22/01/2021 12:13, Nick Altmann wrote:
>
>         It depends on purpose. For example, I need an initial
>         (untouched) packet there.
>
>
>         --
>
>         Nick
>
>          
>
>         пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be
>         <mailto:johan at democon.be>>:
>
>             Hi,
>
>
>             when you use the siptrace module and you go to homer from
>             opensips
>             (hence without a port mirror/span port on a switch), when
>             you change
>             from /to header, then the homer trace doesnot reflect the
>             real packet
>             that exits opensips (as we all know, from to header
>             changes are only
>             active when the packet is send out). 
>
>
>             I find this annoying as the purpose of having a tracer is
>             defeated by
>             this : we want to see the real packets with the changes
>             applied.
>
>
>             Therefore : are there more people on the list who are
>             ennoyed by this ?
>             If yes, then I will open a feature request for it.
>
>
>             wkr,
>
>             _______________________________________________
>             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
>
>
> _______________________________________________
> 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  Fri Jan 22 14:50:59 2021
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 22 Jan 2021 16:50:59 +0200
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <bb76e3b3-fa21-a71e-0cf9-65f63581f6fc@democon.be>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
 <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>
 <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>
 <BN7PR10MB259408FAC86EBF02AA21267C95A09@BN7PR10MB2594.namprd10.prod.outlook.com>
 <bb76e3b3-fa21-a71e-0cf9-65f63581f6fc@democon.be>
Message-ID: <a84be28f-cd3c-0633-ef0b-04f8fd2fbdb1@opensips.org>

Hi Johan!

Just as Ben indicated, you should get two HEP packages per call the 
inbound, which is the message unchanged, and the outbound, which is the 
message sent over the network. Basically you should get both messages - 
are you getting only one? Or you're getting two, but duplicates?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 1/22/21 4:36 PM, johan wrote:
> I use this for invite based dialogs.
> 
> trace("hep_id","d","sip");
> 
> I call it just before topology_hiding()
> 
> I don't see from/to header changes.
> 
> Can you check on your side Ben ?
> 
> 
> wkr,
> 
> On 22/01/2021 15:28, Ben Newlin wrote:
>>
>> We are using siptrace and we get both the incoming and outbound 
>> message captured, with all changes. I think it depends on the scope 
>> you provide [1]. We use scope T for transactions and we get both.
>>
>> Although, we are using OpenSIPS 2.4. It’s possible the functionality 
>> has changed in 3.x. In which case, I would agree that it is not good.
>>
>> [1] - https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace 
>> <https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace>
>>
>> Ben Newlin
>>
>> *From: *Users <users-bounces at lists.opensips.org> on behalf of Andy 
>> Dierlam <adierlam at ptgi-ics.com>
>> *Date: *Friday, January 22, 2021 at 9:13 AM
>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject: *Re: [OpenSIPS-Users] feasablity / interest in a feature 
>> request.
>>
>> How about running a continuous tshark/pcap/ngrep on interfaces of 
>> interest.
>>
>> you can run every 10min.  putting in 10min files.
>>
>> On Fri, Jan 22, 2021 at 6:36 AM johan <johan at democon.be 
>> <mailto:johan at democon.be>> wrote:
>>
>>     yeah me too.
>>
>>     but I want also to see the forwarded packet with all changes.
>>
>>     On 22/01/2021 12:13, Nick Altmann wrote:
>>
>>         It depends on purpose. For example, I need an initial
>>         (untouched) packet there.
>>
>>
>>         --
>>
>>         Nick
>>
>>         пт, 22 янв. 2021 г. в 09:18, johan <johan at democon.be
>>         <mailto:johan at democon.be>>:
>>
>>             Hi,
>>
>>
>>             when you use the siptrace module and you go to homer from
>>             opensips
>>             (hence without a port mirror/span port on a switch), when
>>             you change
>>             from /to header, then the homer trace doesnot reflect the
>>             real packet
>>             that exits opensips (as we all know, from to header
>>             changes are only
>>             active when the packet is send out).
>>
>>
>>             I find this annoying as the purpose of having a tracer is
>>             defeated by
>>             this : we want to see the real packets with the changes
>>             applied.
>>
>>
>>             Therefore : are there more people on the list who are
>>             ennoyed by this ?
>>             If yes, then I will open a feature request for it.
>>
>>
>>             wkr,
>>
>>             _______________________________________________
>>             Users mailing list
>>             Users at lists.opensips.org <mailto: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  <mailto: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 <mailto: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
> 


From razvan at opensips.org  Fri Jan 22 14:57:04 2021
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 22 Jan 2021 16:57:04 +0200
Subject: [OpenSIPS-Users] Siprec contact header
In-Reply-To: <CAMo3uQwM8-B6w-wSzB1q1oZ0tSt_qjDr54emkEREv9fsB3Wd3Q@mail.gmail.com>
References: <CAMo3uQwM8-B6w-wSzB1q1oZ0tSt_qjDr54emkEREv9fsB3Wd3Q@mail.gmail.com>
Message-ID: <bb36b20b-4c52-3630-f9d2-659f50680eab@opensips.org>

Hi, Jofi!

Unfortunately this is not implemented in OpenSIPS right now. Can you 
please open a bug report for this?
In the meantime, as a workaround, you can "catch" the INVITE in the 
local_route and modify the contact accordingly.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 1/21/21 6:24 PM, jofi Y wrote:
> Hello All,
> 
> I have a question regarding contact header generated by siprec module.
> According to rfc7866 (6.1.1):
> 
>    The SRC MUST include the "+sip.src" feature tag in the Contact URI,
>     defined in this specification as an extension to [RFC3840  <https://tools.ietf.org/html/rfc3840>], for all
>     RSs.
> 
> 
> The contact parameter "+sip.src" is not generated when I started recording.
> Currently I use Opensips version 2.4.6 and 3.1.1.
> 
> Is there an alternative way to generate "+sip.src" when recording 
> session started?
> 
> Many thanks,
> Jofi
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 


From Ben.Newlin at genesys.com  Fri Jan 22 15:16:12 2021
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Fri, 22 Jan 2021 15:16:12 +0000
Subject: [OpenSIPS-Users] feasablity / interest in a feature request.
In-Reply-To: <bb76e3b3-fa21-a71e-0cf9-65f63581f6fc@democon.be>
References: <5e434505-37ad-cf8c-826b-4f447360a471@democon.be>
 <CADKNvCqqCqVnyPbrjdTchYZ+2W+ZfMOYKY4kZC13mSeG1LovHw@mail.gmail.com>
 <e0d4a056-f384-0f39-4214-d73a998c8e91@democon.be>
 <CAKe=zzgQRV5M+=tQhhKRd_Ry4m3NmJ5YV8Oyo22-uoih8w8dxA@mail.gmail.com>
 <BN7PR10MB259408FAC86EBF02AA21267C95A09@BN7PR10MB2594.namprd10.prod.outlook.com>,
 <bb76e3b3-fa21-a71e-0cf9-65f63581f6fc@democon.be>
Message-ID: <BN7PR10MB2594769E27D3DCA3E7C3615495A09@BN7PR10MB2594.namprd10.prod.outlook.com>

Johan,

Unfortunately, the server on which I am tracing is only transaction-aware and does not have Dialog, so I cannot verify this.

I will say that we call trace() very early in message processing, almost first thing, and we do still see all changes on the outbound message.

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of johan <johan at democon.be>
Date: Friday, January 22, 2021 at 9:38 AM
To: users at lists.opensips.org <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] feasablity / interest in a feature request.

I use this for invite based dialogs.

trace("hep_id","d","sip");

I call it just before topology_hiding()

I don't see from/to header changes.

Can you check on your side Ben ?



wkr,
On 22/01/2021 15:28, Ben Newlin wrote:
We are using siptrace and we get both the incoming and outbound message captured, with all changes. I think it depends on the scope you provide [1]. We use scope T for transactions and we get both.

Although, we are using OpenSIPS 2.4. It’s possible the functionality has changed in 3.x. In which case, I would agree that it is not good.

[1] - https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace<https://opensips.org/docs/modules/3.1.x/tracer.html#func_trace>

Ben Newlin

From: Users <users-bounces at lists.opensips.org><mailto:users-bounces at lists.opensips.org> on behalf of Andy Dierlam <adierlam at ptgi-ics.com><mailto:adierlam at ptgi-ics.com>
Date: Friday, January 22, 2021 at 9:13 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] feasablity / interest in a feature request.
How about running a continuous tshark/pcap/ngrep on interfaces of interest.
you can run every 10min.  putting in 10min files.

On Fri, Jan 22, 2021 at 6:36 AM johan <johan at democon.be<mailto:johan at democon.be>> wrote:

yeah me too.

but I want also to see the forwarded packet with all changes.
On 22/01/2021 12:13, Nick Altmann wrote:
It depends on purpose. For example, I need an initial (untouched) packet there.

--
Nick

пт, 22 нв. 2021 г. в 09:18, johan <johan at democon.be<mailto:johan at democon.be>>:
Hi,


when you use the siptrace module and you go to homer from opensips
(hence without a port mirror/span port on a switch), when you change
from /to header, then the homer trace doesnot reflect the real packet
that exits opensips (as we all know, from to header changes are only
active when the packet is send out).


I find this annoying as the purpose of having a tracer is defeated by
this : we want to see the real packets with the changes applied.


Therefore : are there more people on the list who are ennoyed by this ?
If yes, then I will open a feature request for it.


wkr,

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

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

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From razvan at opensips.org  Fri Jan 22 15:18:43 2021
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Fri, 22 Jan 2021 17:18:43 +0200
Subject: [OpenSIPS-Users] v3.1 Active/Active maintain active calls on
 node failure
In-Reply-To: <orpf0dh32u5n0m49ct01qqth.1611075742840@gmail.com>
References: <orpf0dh32u5n0m49ct01qqth.1611075742840@gmail.com>
Message-ID: <8008a332-2f84-5aa7-53b9-96bd560e3ad7@opensips.org>

Hi, Kevin!

There is a tutorial for 2.4 for this[1] but most of the functions 
changed a bit in 3.1.

Your setup seems correct
* you should have different sharing tags for each node
* for each created dialog, you should set the tag of the current node
* you should setup dialog replication between the two nodes
* you must figure out when a node goes down (probably using external 
tools) - once you figure out, you should change the active state of the 
tag on the remaining node - he should be active for both tags (do this 
using the clusterer_shtag_set_active command, as Social Boh indicated)
* since you're using a different IP, you should send re-invites for all 
the affected dialogs (just as you indicated)

Please provide a trace for the messages sent after you switch, to 
understand why the call is being terminated.

[1] 
https://blog.opensips.org/2018/03/23/clustering-ongoing-calls-with-opensips-2-4/

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com

On 1/19/21 7:02 PM, Kevin Vines wrote:
> Hi Social,
> 
> I'll try my luck with Google Translate if you want to share your tutorial.
> 
> Thanks,
> 
> Kevin V.
> 
> *From:* users at lists.opensips.org
> *Sent:* January 19, 2021 11:03 a.m.
> *To:* kworm at missouri-telecom.com; users at lists.opensips.org
> *Reply to:* social at bohboh.info; users at lists.opensips.org
> *Subject:* Re: [OpenSIPS-Users] v3.1 Active/Active maintain active calls 
> on node failure
> 
> 
> I think your best option is KeepAlived; on keepalived configuration you 
> declare a script name where you execute:
> 
> /usr/local/bin/opensips-cli -x mi clusterer_shtag_set_active vip/3
> 
> to switch VIP TAG from one server to other.
> 
> In this case BYE go to the right place.
> 
> If anyone want translate from spanish to english, I have a complete 
> tutorial for OpenSIPs 3.1
> 
> Regards
> 
> ---
> I'm SoCIaL, MayBe
> 
> El 19/01/2021 a las 10:40 a. m., Kevin Wormington escribió:
>> I’m not using RTPEngine…the upstream proxies are handling all media, 
>> NAT traversal, etc. so the OpenSIPS instances can always reach the 
>> endpoints.  I’m using clusterer module to share the user location and 
>> dialogs with different active tags per node.  There is zero loss of 
>> media on switch-over and sometimes a little longer PDD for new calls 
>> during switchover until the upstream proxies detect the instance down. 
>>   The only part I can’t seem to get to work is handling the final BYE 
>> for calls that were on the failed node originally.   The re-invite 
>> ping will correct end them but would like to be able to fix it 
>> completely…but maybe that is not currently possible.
>>
>>
>> Thanks,
>>
>> Kevin
>>> On Jan 19, 2021, at 9:31 AM, Social Boh via Users 
>>> <users at lists.opensips.org <mailto:users at lists.opensips.org>> wrote:
>>>
>>> To switch calls from one server to another you have to use redis and 
>>> rptengine using HA with pacemaker y corosync.
>>>
>>> You must have two OpenSIPs, Two RTPEngine, Two Redis servers 
>>> (primary-replica) Two Mariad  servers (primary/primary)
>>>
>>> With redis you can save calls data (ip, ports, callid) on active 
>>> server and then use these data on the replica server when swithc to 
>>> active. On my tests, when switching from a server to another I have 
>>> between 5 and 10 seconds without audio.
>>>
>>> Regards
>>>
>>> ---
>>> I'm SoCIaL, MayBe
>>>
>>> El 19/01/2021 a las 10:00 a. m., Kevin Wormington escribió:
>>>> I’m not using a VIP and I have made some progress by setting a 
>>>> different active tag on each node…then upon node failure setting the 
>>>> failed node's tag to active on remaining node.  This lets the 
>>>> re-invite pinging work, etc.  It’s almost there but the handling of 
>>>> the BYE…they are still sent to the IP of the failed node even after 
>>>> re-invite pings so any in-progress calls from the failed node are 
>>>> zombie when they hang up until the re-invite ping times out (30 
>>>> seconds).   I found an article about initiating a re-invite on the 
>>>> new node with something like "opensips-cli -x mi dlg_send_sequential 
>>>> callid="442CB6C1-6005F8B80009DA08-FC731700" mode=challenge 
>>>> body=outbound” but that either seems to terminate the call 
>>>> immediately or say the dialog wasn’t found.
>>>>
>>>>
>>>> Thanks,
>>>>
>>>> Kevin
>>>>> On Jan 19, 2021, at 8:46 AM, Andy Dierlam <adierlam at ptgi-ics.com 
>>>>> <mailto:adierlam at ptgi-ics.com>> wrote:
>>>>>
>>>>> With dialog writing to db that both servers use.   And same tag on 
>>>>> both - modparam("dialog", "dlg_sharing_tag", "vip1=active")
>>>>> had this working on opensips 2.4
>>>>>
>>>>> thanks
>>>>> Andy
>>>>>
>>>>>
>>>>> On Mon, Jan 18, 2021 at 2:30 PM Kevin Wormington 
>>>>> <kworm at missouri-telecom.com <mailto:kworm at missouri-telecom.com>> wrote:
>>>>> Hi,
>>>>>
>>>>> I've been attempting to get a two node active/active setup to work 
>>>>> with the v3.1 clusterer module sharing usrloc and dialog.  The 
>>>>> setup is fronted by a proxy that handles all of the NAT/media so 
>>>>> either OpenSIPS instance can communicate directly with the user.
>>>>>
>>>>> What I have working so far:
>>>>>
>>>>> Registrations and calls work when sent to either node and if you 
>>>>> stop OpenSIPS on a node new calls work fine using the other node.
>>>>>
>>>>> What I can’t get to work:
>>>>>
>>>>> Calls that are already in progress to switch between nodes when one 
>>>>> node fails.
>>>>>
>>>>>
>>>>> I have messed around with various sharing tags…no tag, same tag, 
>>>>> different tags but haven’t had any luck.   I’m guessing that I’m 
>>>>> missing something to trigger the remaining node to send re-invites. 
>>>>>  Has anyone attempted this type of setup and have any ideas?
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Kevin
>>>>> _______________________________________________
>>>>> 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
>>>>
>>>> _______________________________________________
>>>> 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
> 


From alexanderhenryperkins at gmail.com  Sun Jan 24 01:37:59 2021
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Sat, 23 Jan 2021 20:37:59 -0500
Subject: [OpenSIPS-Users] rest_client and basic authentication
Message-ID: <CALLkTp16AhE0UY2zZ+1_MgqsmXryCf-B90hdE-z53UkojZJyNg@mail.gmail.com>

HI All. I need to consume a RESTful web service and pass that information
to avp variables.  I was looking at the rest_client module, but cannot find
how to authenticate using basic authentication.  Does anyone have
suggestions?

Thank you,
Alex
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From tpaivaa at gmail.com  Sun Jan 24 06:42:55 2021
From: tpaivaa at gmail.com (Tomi Hakkarainen)
Date: Sun, 24 Jan 2021 08:42:55 +0200
Subject: [OpenSIPS-Users] rest_client and basic authentication
In-Reply-To: <CALLkTp16AhE0UY2zZ+1_MgqsmXryCf-B90hdE-z53UkojZJyNg@mail.gmail.com>
References: <CALLkTp16AhE0UY2zZ+1_MgqsmXryCf-B90hdE-z53UkojZJyNg@mail.gmail.com>
Message-ID: <B983F3B3-93F4-414C-976B-9BB3285C40DC@gmail.com>

Hi,

it seems that you should just create the header by your self and add it before reguests. 
rest_append_hf("Authorization: Basic <BASE64 of creds here>");

when making a request. In basic HTTP authentication, a request contains a header field in the form of Authorization: Basic <credentials>, where credentials is the Base64 encoding of ID and password joined by a single colon :.


Tomi

On 24. Jan 2021, at 3.40, Alexander Perkins <alexanderhenryperkins at gmail.com> wrote:


HI All. I need to consume a RESTful web service and pass that information to avp variables.  I was looking at the rest_client module, but cannot find how to authenticate using basic authentication.  Does anyone have suggestions?

Thank you,
Alex
_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From rosenberg11219 at gmail.com  Sun Jan 24 06:56:04 2021
From: rosenberg11219 at gmail.com (Schneur Rosenberg)
Date: Sun, 24 Jan 2021 08:56:04 +0200
Subject: [OpenSIPS-Users] Limit Call per Second
In-Reply-To: <CAND0LksvhjUzTxh9F9Q-rMF8sB_exD_HSJ70x3mw+0t=wWc7_g@mail.gmail.com>
References: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
 <CAND0LksvhjUzTxh9F9Q-rMF8sB_exD_HSJ70x3mw+0t=wWc7_g@mail.gmail.com>
Message-ID: <CANvjR0UuKbvhskg6qxPgtr9mem+KV5wOA+kY2mej6tRW1sxGTw@mail.gmail.com>

Maybe use memcache, it's global and it won't create a bottleneck like a
database query would.

Scott (Schneur)

On Thu, Jan 21, 2021, 05:43 Ovidiu Sas <osas at voipembedded.com> wrote:

> Take a look at the ratelimit module:
> https://opensips.org/docs/modules/3.1.x/ratelimit.html
>
> -ovidiu
>
> On Wed, Jan 20, 2021 at 10:01 PM Alexander Perkins
> <alexanderhenryperkins at gmail.com> wrote:
> >
> > Hi All.  Is there a way to limit the calls per second by pulling the
> information from a database? I was looking at the call_control module and
> that seems to be a global value. but I would need something that I can
> control at the call level (I will be handling different accounts that have
> different limits).
> >
> > Thanks, All.  Any help is appreciated.
> >
> > Alex
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From osas at voipembedded.com  Sun Jan 24 14:54:16 2021
From: osas at voipembedded.com (Ovidiu Sas)
Date: Sun, 24 Jan 2021 09:54:16 -0500
Subject: [OpenSIPS-Users] Limit Call per Second
In-Reply-To: <CANvjR0UuKbvhskg6qxPgtr9mem+KV5wOA+kY2mej6tRW1sxGTw@mail.gmail.com>
References: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
 <CAND0LksvhjUzTxh9F9Q-rMF8sB_exD_HSJ70x3mw+0t=wWc7_g@mail.gmail.com>
 <CANvjR0UuKbvhskg6qxPgtr9mem+KV5wOA+kY2mej6tRW1sxGTw@mail.gmail.com>
Message-ID: <CAND0LktzCxcvi80dfvScheybq4z=hpJ3LoUXZdgXRxh09iqTZQ@mail.gmail.com>

There are no db queries, unless explicitly enabled for distributed
rate limiting, in which case memcache or redis can be used.

-ovidiu

On Sun, Jan 24, 2021 at 1:57 AM Schneur Rosenberg
<rosenberg11219 at gmail.com> wrote:
>
> Maybe use memcache, it's global and it won't create a bottleneck like a database query would.
>
> Scott (Schneur)
>
> On Thu, Jan 21, 2021, 05:43 Ovidiu Sas <osas at voipembedded.com> wrote:
>>
>> Take a look at the ratelimit module:
>> https://opensips.org/docs/modules/3.1.x/ratelimit.html
>>
>> -ovidiu
>>
>> On Wed, Jan 20, 2021 at 10:01 PM Alexander Perkins
>> <alexanderhenryperkins at gmail.com> wrote:
>> >
>> > Hi All.  Is there a way to limit the calls per second by pulling the information from a database? I was looking at the call_control module and that seems to be a global value. but I would need something that I can control at the call level (I will be handling different accounts that have different limits).
>> >
>> > Thanks, All.  Any help is appreciated.
>> >
>> > Alex
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.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



-- 
VoIP Embedded, Inc.
http://www.voipembedded.com


From jofi.yance at gmail.com  Mon Jan 25 07:19:07 2021
From: jofi.yance at gmail.com (jofi Y)
Date: Mon, 25 Jan 2021 08:19:07 +0100
Subject: [OpenSIPS-Users] Siprec contact header
In-Reply-To: <CAMo3uQwM8-B6w-wSzB1q1oZ0tSt_qjDr54emkEREv9fsB3Wd3Q@mail.gmail.com>
References: <CAMo3uQwM8-B6w-wSzB1q1oZ0tSt_qjDr54emkEREv9fsB3Wd3Q@mail.gmail.com>
Message-ID: <CAMo3uQz_=zWVTCDy6Gd1WHGmvybG+F9QqbyR9MQ+3FAW6LxOqQ@mail.gmail.com>

Hi  Răzvan,

Thank you for your feedback.
I opened bug report #2383 on Github.

With regards,
Jofi

Date: Fri, 22 Jan 2021 16:57:04 +0200
From: Răzvan Crainea <razvan at opensips.org>
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] Siprec contact header
Message-ID: <bb36b20b-4c52-3630-f9d2-659f50680eab at opensips.org>
Content-Type: text/plain; charset=utf-8; format=flowed

Hi, Jofi!

Unfortunately this is not implemented in OpenSIPS right now. Can you
please open a bug report for this?
In the meantime, as a workaround, you can "catch" the INVITE in the
local_route and modify the contact accordingly.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
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From krishdinesh at yahoo.com  Mon Jan 25 09:08:49 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Mon, 25 Jan 2021 09:08:49 +0000 (UTC)
Subject: [OpenSIPS-Users] REST / rest_get Challenge ...
References: <1605769356.4889361.1611565729925.ref@mail.yahoo.com>
Message-ID: <1605769356.4889361.1611565729925@mail.yahoo.com>

Hello,
I am trying to test a REST API call using the example code provided in the documentation. I am experiencing the following error and opensips would not start. I am running opensips 3.1.1 (x86_64/linux) on Debian 10. Any help would be appreciated. Curl is also installed and i am able to query a URL successfully Thank you
...# Example of querying a REST service to get the credit of an account$var(rc) = rest_get("https://getcredit.org/?account=$fU",                    $var(credit),                    $var(ct),                    $var(rcode));if ($var(rc) < 0) { xlog("rest_get() failed with $var(rc), acc=$fU\n"); send_reply(500, "Server Internal Error"); exit;}
if ($var(rcode) != 200) { xlog("L_INFO", "rest_get() rcode=$var(rcode), acc=$fU\n"); send_reply(403, "Forbidden"); exit;}


Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: INFO:rest_client:mod_init: Module initialized!Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: INFO:auth:mod_init: initializing...Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_cmd: Param [2] expected to be a variableJan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: Failed to fix command <rest_get>Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: fixing failed (code=-6) at /etc/opensips/opensips.cfg:232Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: CRITICAL:core:fix_expr: fix_actions errorJan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:main: failed to fix configuration with err code -6
Thank you,Dinesh
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From liviu at opensips.org  Mon Jan 25 09:13:36 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 25 Jan 2021 11:13:36 +0200
Subject: [OpenSIPS-Users] REST / rest_get Challenge ...
In-Reply-To: <1605769356.4889361.1611565729925@mail.yahoo.com>
References: <1605769356.4889361.1611565729925.ref@mail.yahoo.com>
 <1605769356.4889361.1611565729925@mail.yahoo.com>
Message-ID: <ea8455f7-e8b5-64b6-52cf-3dab5dcbd589@opensips.org>

On 25.01.2021 11:08, Dinesh Krishnamurthy via Users wrote:
> Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_cmd: 
> Param [2] expected to be a variable
> Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: 
> ERROR:core:fix_actions: Failed to fix command <rest_get>
> Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: 
> ERROR:core:fix_actions: fixing failed (code=-6) at 
> /etc/opensips/opensips.cfg:232
> Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: 
> CRITICAL:core:fix_expr: fix_actions error

Hi,

You are calling rest_get() with a quoted 2nd parameter, but the function 
expects a non-quoted variable.  The example snippet you gave worked well 
on my test, so I suspect there is another rest_get() call which is 
incorrect, see line 232 in your file.

Regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com

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From liviu at opensips.org  Mon Jan 25 09:21:34 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 25 Jan 2021 11:21:34 +0200
Subject: [OpenSIPS-Users] Rate_Cacher
In-Reply-To: <1610959294662-0.post@n2.nabble.com>
References: <1610959294662-0.post@n2.nabble.com>
Message-ID: <f89c9518-e692-c11e-0719-1b58897bcbb5@opensips.org>

On 18.01.2021 10:41, zozon wrote:
> the functionality of rate_cacher would be enough for me. therefore I was wondering, when the
> beta/production version could be expected. Any milestones you could share?

Hi,

I've just updated rate_cacher's status from "alpha" to "beta", as it 
passed the OpenSIPS 3.1 beta
testing phase, after all.

At the time when I reviewed the rate cacher Pull Request, the code 
seemed quite healthy.  Moreover,
let's not forget that it's likely running in production as we're 
speaking right now.

So please give it a spin and report any issues you may run into on 
GitHub - one of the devs will
likely help you out!

Regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com



From callum.guy at x-on.co.uk  Mon Jan 25 09:23:47 2021
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Mon, 25 Jan 2021 09:23:47 +0000
Subject: [OpenSIPS-Users] Limit Call per Second
In-Reply-To: <CAND0LktzCxcvi80dfvScheybq4z=hpJ3LoUXZdgXRxh09iqTZQ@mail.gmail.com>
References: <CALLkTp3+zuraEaNfvP-NRpboe+w92zvTwc3wzoQm24z3NjOgjQ@mail.gmail.com>
 <CAND0LksvhjUzTxh9F9Q-rMF8sB_exD_HSJ70x3mw+0t=wWc7_g@mail.gmail.com>
 <CANvjR0UuKbvhskg6qxPgtr9mem+KV5wOA+kY2mej6tRW1sxGTw@mail.gmail.com>
 <CAND0LktzCxcvi80dfvScheybq4z=hpJ3LoUXZdgXRxh09iqTZQ@mail.gmail.com>
Message-ID: <CAFjCFznxELGUc1evL_uBMQ2TV0cUwboctokT8qExJM-eHnppqA@mail.gmail.com>

An alternative option would be to leverage cachedb_local and opensips-cli
to implement your list of accounts and rate limits. It has the advantage of
using the internal opensips cache service and is probably your most high
performance option, with the CLI you can automate data refreshes using
basic cron scripts or anything you fancy!

https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4
https://www.opensips.org/Documentation/Interface-CoreMI-3-0#toc15
https://www.opensips.org/Documentation/Tutorials-KeyValueInterface

On Sun, 24 Jan 2021 at 14:56, Ovidiu Sas <osas at voipembedded.com> wrote:

> There are no db queries, unless explicitly enabled for distributed
> rate limiting, in which case memcache or redis can be used.
>
> -ovidiu
>
> On Sun, Jan 24, 2021 at 1:57 AM Schneur Rosenberg
> <rosenberg11219 at gmail.com> wrote:
> >
> > Maybe use memcache, it's global and it won't create a bottleneck like a
> database query would.
> >
> > Scott (Schneur)
> >
> > On Thu, Jan 21, 2021, 05:43 Ovidiu Sas <osas at voipembedded.com> wrote:
> >>
> >> Take a look at the ratelimit module:
> >> https://opensips.org/docs/modules/3.1.x/ratelimit.html
> >>
> >> -ovidiu
> >>
> >> On Wed, Jan 20, 2021 at 10:01 PM Alexander Perkins
> >> <alexanderhenryperkins at gmail.com> wrote:
> >> >
> >> > Hi All.  Is there a way to limit the calls per second by pulling the
> information from a database? I was looking at the call_control module and
> that seems to be a global value. but I would need something that I can
> control at the call level (I will be handling different accounts that have
> different limits).
> >> >
> >> > Thanks, All.  Any help is appreciated.
> >> >
> >> > Alex
> >> > _______________________________________________
> >> > Users mailing list
> >> > Users at lists.opensips.org
> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >>
> >>
> >>
> >> --
> >> VoIP Embedded, Inc.
> >> http://www.voipembedded.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
>
>
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>

-- 





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From krishdinesh at yahoo.com  Mon Jan 25 09:25:52 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Mon, 25 Jan 2021 09:25:52 +0000 (UTC)
Subject: [OpenSIPS-Users] REST / rest_get Challenge ...
In-Reply-To: <ea8455f7-e8b5-64b6-52cf-3dab5dcbd589@opensips.org>
References: <1605769356.4889361.1611565729925.ref@mail.yahoo.com>
 <1605769356.4889361.1611565729925@mail.yahoo.com>
 <ea8455f7-e8b5-64b6-52cf-3dab5dcbd589@opensips.org>
Message-ID: <1634036332.2457112.1611566752363@mail.yahoo.com>

 Hi,
I am using the exact string as in example provided in the link below. If there any dependency that could cause this issue ? Thank you,  
Example 1.8. rest_get usage
https://opensips.org/html/docs/modules/2.3.x/rest_client.html


Thank you,Dinesh
    On Monday, January 25, 2021, 02:45:24 PM GMT+5:30, Liviu Chircu <liviu at opensips.org> wrote:  
 
  On 25.01.2021 11:08, Dinesh Krishnamurthy via Users wrote:
  
 Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_cmd: Param [2] expected to be a variable Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: Failed to fix command <rest_get> Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: fixing failed (code=-6) at /etc/opensips/opensips.cfg:232 Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: CRITICAL:core:fix_expr: fix_actions error 
 
Hi,
 
You are calling rest_get() with a quoted 2nd parameter, but the function expects a non-quoted variable.  The example snippet you gave worked well on my test, so I suspect there is another rest_get() call which is incorrect, see line 232 in your file.
 
Regards,
 
 -- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  
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From liviu at opensips.org  Mon Jan 25 09:29:55 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Mon, 25 Jan 2021 11:29:55 +0200
Subject: [OpenSIPS-Users] REST / rest_get Challenge ...
In-Reply-To: <1634036332.2457112.1611566752363@mail.yahoo.com>
References: <1605769356.4889361.1611565729925.ref@mail.yahoo.com>
 <1605769356.4889361.1611565729925@mail.yahoo.com>
 <ea8455f7-e8b5-64b6-52cf-3dab5dcbd589@opensips.org>
 <1634036332.2457112.1611566752363@mail.yahoo.com>
Message-ID: <18e412bc-55f8-952f-1456-c478ebdc2fb7@opensips.org>

On 25.01.2021 11:25, Dinesh Krishnamurthy via Users wrote:
> I am using the exact string as in example provided in the link below. 
> If there any dependency that could cause this issue ? Thank you,
> *
> *
> *Example 1.8. |rest_get| usage*
>
> https://opensips.org/html/docs/modules/2.3.x/rest_client.html

You are copy-pasting from the OpenSIPS 2.3 docs, not 3.1. The syntax has 
changed in the meantime! ;)

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com

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From krishdinesh at yahoo.com  Mon Jan 25 12:43:30 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Mon, 25 Jan 2021 12:43:30 +0000 (UTC)
Subject: [OpenSIPS-Users] REST / rest_get Challenge ...
In-Reply-To: <18e412bc-55f8-952f-1456-c478ebdc2fb7@opensips.org>
References: <1605769356.4889361.1611565729925.ref@mail.yahoo.com>
 <1605769356.4889361.1611565729925@mail.yahoo.com>
 <ea8455f7-e8b5-64b6-52cf-3dab5dcbd589@opensips.org>
 <1634036332.2457112.1611566752363@mail.yahoo.com>
 <18e412bc-55f8-952f-1456-c478ebdc2fb7@opensips.org>
Message-ID: <1959239527.4933286.1611578610475@mail.yahoo.com>

 Thank you Liviu. Yes i realized the problem with the syntax and made the change. It is working now.
I had to search for rest_get function and was directed to the earlier version of the documentation which i followed. 
    On Monday, January 25, 2021, 03:01:38 PM GMT+5:30, Liviu Chircu <liviu at opensips.org> wrote:  
 
  On 25.01.2021 11:25, Dinesh Krishnamurthy via Users wrote:
  
 I am using the exact string as in example provided in the link below. If there any dependency that could cause this issue ? Thank you,   
  Example 1.8. rest_get usage 
  https://opensips.org/html/docs/modules/2.3.x/rest_client.html 
 
You are copy-pasting from the OpenSIPS 2.3 docs, not 3.1.  The syntax has changed in the meantime! ;)

  -- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com _______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
  
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From liviu at opensips.org  Tue Jan 26 05:03:05 2021
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 26 Jan 2021 07:03:05 +0200
Subject: [OpenSIPS-Users] Question regarding Federated User Location
 Cluster
In-Reply-To: <tencent_2BDF660DD375931304A79016DD3F8B7DDE06@qq.com>
References: <tencent_2BDF660DD375931304A79016DD3F8B7DDE06@qq.com>
Message-ID: <2c02f27f-a284-93f0-8a15-6d1b4b82c6e9@opensips.org>

On 22.01.2021 04:57, Jeffrey Zhao wrote:
>
> For example, for formal production system senario, two sites, with two 
> opensips nodes for each site, HA mode for each site.
> 1. Should I deploy MySQL and Cassandra on each node? 4 MySQL instances 
> and 4 Cassandra on each node?
> 2. Should I setup db replication among 4 MySQL instances or just 
> standalone separated setup?
> 3. Should I put all 4  Cassandra instances into one cluster?
> 4. For each HA virtual IP setup, what's the recommended tool, keepalived?

Hi, Jeffrey!

Those are some excellent questions!  Let's drill down into each of them, 
as I will also add them into an "FAQ" section towards the end of the 
tutorial:

1. For MySQL: yes, because the main goal is to have each instance able 
to run on its own.  For Cassandra, you can build the shared cluster 
using any configuration you prefer.  Single node, 3 nodes, 5... the only 
requirement is for all locations to be able to connect to it.

2. No MySQL replication!  The MySQL is simply used for restart 
persistence purposes.

3. Yes, you must.  There is no purpose for a stand-alone Cassandra in 
this architecture.

4. Personally, I have nothing but good things to say for both 
"keepalived" and its older brother from another mother, "vrrpd".

Cheers,

-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com



From mark at allenclan.co.uk  Tue Jan 26 10:47:51 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Tue, 26 Jan 2021 10:47:51 +0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
In-Reply-To: <PR1PR06MB4683C68851C095F4CB03C9D8A7A10@PR1PR06MB4683.eurprd06.prod.outlook.com>
References: <001c01d6efd9$71d7fb50$5587f1f0$@smartvox.co.uk>
 <PR1PR06MB4683C68851C095F4CB03C9D8A7A10@PR1PR06MB4683.eurprd06.prod.outlook.com>
Message-ID: <CADaqb1vczxMELybNNOj1hvw_6WYD42DfF9pvjsN2JpoGHKpLuw@mail.gmail.com>

Hi John and Johan - thanks for your replies. I'll have a look at RTPEngine
to see if it makes things simpler for me.

I have managed to get audio working both ways with Mediaproxy - the problem
I was encountering was with config.ini settings. I had to explicitly set
"relay_ip" and restarted Mediaproxy relay, dispatcher, and OpenSIPS after
which audio worked both ways.
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From jeff at ugnd.org  Tue Jan 26 15:34:59 2021
From: jeff at ugnd.org (Jeff Pyle)
Date: Tue, 26 Jan 2021 10:34:59 -0500
Subject: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong RURI
	domain on sequential requests
Message-ID: <CAPhW+0KkarPTe9sQjyYFUu5NuZc3rvLX0O4EqMa2-9pB45-yYw@mail.gmail.com>

Hello,

This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1.

I have a registrar configured with basic call routing between the
registered AORs.  I use topology_hiding("D") to create the dialog on calls
and normal stuff like has_totag() and topology_hiding_match() for
sequential request handling.  All this seems fine.

This appears high in the main route and appears to do exactly what it
should:

        if (has_body("application/sdp")) {
                if (nat_uac_test(14)) {
                        setflag("NAT_FLAG");
                }
        } else {
                if (nat_uac_test(6)) {
                        setflag("NAT_FLAG");
                }
        }

        if (isflagset("NAT_FLAG")) {
                force_rport();
                if ($rm == "REGISTER") {
                        fix_nated_register();
                } else {
                        fix_nated_contact();
                }
        }

And, for replies:

onreply_route [handle_rtprelay_onreply] {
        # rtpengine and such, omitted for brevity
        if (isbflagset("NAT_BFLAG")) {
                fix_nated_contact();
        }

        exit;
}

When one client calls another, everything works fine.  lookup("location")
works to update $rd with the original (private) Contact provided upon
registration, and $du contains the actual received source IP:port to get to
the device.  Excellent.  The INVITE goes out accordingly, and all is well.

My problem occurs with sequential requests, say, re-INVITEs from on-hold
events.  The dialogs themselves save the received IP:port values as the
caller_contact and callee_contact values (from fix_nated_contact() above),
so when the requests pass through the sequential handling section of the
script and topology_hiding_match() does its fixups, the request URI domain
of the relayed request has the received IP:port values of the target UA
rather than the private IP:port values the UA provided during the initial
request that established the dialog.

I can't wrap my head around how to fix this.  The initial requests work
because lookup() has the intelligence to distinguish the UAC's Contact from
the received IP:port at REGISTER-time, but I can't see how to achieve this
at dialog-creation time so sequential requests have the right RURI domain.
Force the caller_contact and callee_contact to the private values somehow,
and manage the route_set to point to the appropriate received IP:port?  I'm
not sure how to configure that if it is the solution.

Any direction would be appreciated!


Regards,
Jeff
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From mark at allenclan.co.uk  Tue Jan 26 15:47:45 2021
From: mark at allenclan.co.uk (Mark Allen)
Date: Tue, 26 Jan 2021 15:47:45 +0000
Subject: [OpenSIPS-Users] Mediaproxy configuration
In-Reply-To: <CADaqb1vczxMELybNNOj1hvw_6WYD42DfF9pvjsN2JpoGHKpLuw@mail.gmail.com>
References: <001c01d6efd9$71d7fb50$5587f1f0$@smartvox.co.uk>
 <PR1PR06MB4683C68851C095F4CB03C9D8A7A10@PR1PR06MB4683.eurprd06.prod.outlook.com>
 <CADaqb1vczxMELybNNOj1hvw_6WYD42DfF9pvjsN2JpoGHKpLuw@mail.gmail.com>
Message-ID: <CADaqb1vmQ88-nsB9XfVx1qQ2fGVYiEe+=wU4eXv+P1GqW-H8Dg@mail.gmail.com>

Further to this - as I said the relay_ip overcame the immediate audio
problem, but on testing it timed out after just over 60 seconds. Looking at
the traffic in Wireshark and the SDP in SIP messages the cause seems to be
that Asterisk is sending RTP direct to the 46.xxx.xxx.xxx address rather
than via the relay, while traffic in the other direction is coming via the
relay - so after about a minute Mediaproxy thinks one end is dead and
aborts the connection.

This is obviously the issue you flagged up John where you said "You need
the address in the SDP to be the public IP 4x.xxx.xxx.xxx when it
is sending packets to the UAC but you need it to use its LAN address when
sending to the Asterisk server."

Looks like I'll have to use RTPEngine bridging mode instead. Thanks for the
help again :)
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From Johan at democon.be  Tue Jan 26 16:03:04 2021
From: Johan at democon.be (Johan De Clercq)
Date: Tue, 26 Jan 2021 17:03:04 +0100
Subject: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong
 RURI domain on sequential requests
In-Reply-To: <CAPhW+0KkarPTe9sQjyYFUu5NuZc3rvLX0O4EqMa2-9pB45-yYw@mail.gmail.com>
References: <CAPhW+0KkarPTe9sQjyYFUu5NuZc3rvLX0O4EqMa2-9pB45-yYw@mail.gmail.com>
Message-ID: <CAEVWGm96CD2Pe5=0AOB5pAQhs8MDXO8OBEOvyMcX3JUsk98WPQ@mail.gmail.com>

did you change the loose route part to fix route dialog ?

Op di 26 jan. 2021 om 16:39 schreef Jeff Pyle <jeff at ugnd.org>:

> Hello,
>
> This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1.
>
> I have a registrar configured with basic call routing between the
> registered AORs.  I use topology_hiding("D") to create the dialog on calls
> and normal stuff like has_totag() and topology_hiding_match() for
> sequential request handling.  All this seems fine.
>
> This appears high in the main route and appears to do exactly what it
> should:
>
>         if (has_body("application/sdp")) {
>                 if (nat_uac_test(14)) {
>                         setflag("NAT_FLAG");
>                 }
>         } else {
>                 if (nat_uac_test(6)) {
>                         setflag("NAT_FLAG");
>                 }
>         }
>
>         if (isflagset("NAT_FLAG")) {
>                 force_rport();
>                 if ($rm == "REGISTER") {
>                         fix_nated_register();
>                 } else {
>                         fix_nated_contact();
>                 }
>         }
>
> And, for replies:
>
> onreply_route [handle_rtprelay_onreply] {
>         # rtpengine and such, omitted for brevity
>         if (isbflagset("NAT_BFLAG")) {
>                 fix_nated_contact();
>         }
>
>         exit;
> }
>
> When one client calls another, everything works fine.  lookup("location")
> works to update $rd with the original (private) Contact provided upon
> registration, and $du contains the actual received source IP:port to get to
> the device.  Excellent.  The INVITE goes out accordingly, and all is well.
>
> My problem occurs with sequential requests, say, re-INVITEs from on-hold
> events.  The dialogs themselves save the received IP:port values as the
> caller_contact and callee_contact values (from fix_nated_contact() above),
> so when the requests pass through the sequential handling section of the
> script and topology_hiding_match() does its fixups, the request URI domain
> of the relayed request has the received IP:port values of the target UA
> rather than the private IP:port values the UA provided during the initial
> request that established the dialog.
>
> I can't wrap my head around how to fix this.  The initial requests work
> because lookup() has the intelligence to distinguish the UAC's Contact from
> the received IP:port at REGISTER-time, but I can't see how to achieve this
> at dialog-creation time so sequential requests have the right RURI domain.
> Force the caller_contact and callee_contact to the private values somehow,
> and manage the route_set to point to the appropriate received IP:port?  I'm
> not sure how to configure that if it is the solution.
>
> Any direction would be appreciated!
>
>
> Regards,
> Jeff
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From jeff at ugnd.org  Tue Jan 26 16:36:32 2021
From: jeff at ugnd.org (Jeff Pyle)
Date: Tue, 26 Jan 2021 11:36:32 -0500
Subject: [OpenSIPS-Users] Dialogs with fix_nated_contact() have wrong
 RURI domain on sequential requests
In-Reply-To: <CAEVWGm96CD2Pe5=0AOB5pAQhs8MDXO8OBEOvyMcX3JUsk98WPQ@mail.gmail.com>
References: <CAPhW+0KkarPTe9sQjyYFUu5NuZc3rvLX0O4EqMa2-9pB45-yYw@mail.gmail.com>
 <CAEVWGm96CD2Pe5=0AOB5pAQhs8MDXO8OBEOvyMcX3JUsk98WPQ@mail.gmail.com>
Message-ID: <CAPhW+0Kw=eT_DDVrR_mxJt0t6OhYh-=w1quhG3+qYHhSXq3huA@mail.gmail.com>

Hi Johan,

There typically isn't loose_route() in my script because there
is topology_hiding_match() instead.  But, I've tested without topology
hiding (using loose_route for sequential requests) and there is no
difference.

The docs for fix_route_dialog()
<https://opensips.org/html/docs/modules/3.1.x/dialog.html#func_fix_route_dialog>
say that it "forces an in dialog SIP message to contain the ruri, route
headers and dst_uri, as specified by the internal data of the dialog it
belongs to."  That's not a problem here; the in-dialog request already has
the same values as the internal data of the dialog it belongs to.  This
function looks more to prevent bad actors from doing nasty things in
in-dialog requests.  In my case everyone is playing by the rules.

The caller_contact and callee_contact from the "internal data of the
dialog" (as viewed with the dlg_list MI command) contain the
public/received IP and port rather than the internal/private IP and port
each UA provided.  That occurs because of the fix_nated_contact() function
in the script prior to dialog creation.  In other words, by the time the
dialog is created, the internal IP:port is lost.

My questions are:
 - how to preserve the private/internal Contact info in the dialog, and
 - use it for signaling in the RURI but continue to use the received/public
info for routing for in-dialog requests


- Jeff

On Tue, Jan 26, 2021 at 11:04 AM Johan De Clercq <Johan at democon.be> wrote:

> did you change the loose route part to fix route dialog ?
>
> Op di 26 jan. 2021 om 16:39 schreef Jeff Pyle <jeff at ugnd.org>:
>
>> Hello,
>>
>> This is on OpenSIPS nightly 3.1.1~20210125~8bab0da7b-1.
>>
>> I have a registrar configured with basic call routing between the
>> registered AORs.  I use topology_hiding("D") to create the dialog on calls
>> and normal stuff like has_totag() and topology_hiding_match() for
>> sequential request handling.  All this seems fine.
>>
>> This appears high in the main route and appears to do exactly what it
>> should:
>>
>>         if (has_body("application/sdp")) {
>>                 if (nat_uac_test(14)) {
>>                         setflag("NAT_FLAG");
>>                 }
>>         } else {
>>                 if (nat_uac_test(6)) {
>>                         setflag("NAT_FLAG");
>>                 }
>>         }
>>
>>         if (isflagset("NAT_FLAG")) {
>>                 force_rport();
>>                 if ($rm == "REGISTER") {
>>                         fix_nated_register();
>>                 } else {
>>                         fix_nated_contact();
>>                 }
>>         }
>>
>> And, for replies:
>>
>> onreply_route [handle_rtprelay_onreply] {
>>         # rtpengine and such, omitted for brevity
>>         if (isbflagset("NAT_BFLAG")) {
>>                 fix_nated_contact();
>>         }
>>
>>         exit;
>> }
>>
>> When one client calls another, everything works fine.  lookup("location")
>> works to update $rd with the original (private) Contact provided upon
>> registration, and $du contains the actual received source IP:port to get to
>> the device.  Excellent.  The INVITE goes out accordingly, and all is well.
>>
>> My problem occurs with sequential requests, say, re-INVITEs from on-hold
>> events.  The dialogs themselves save the received IP:port values as the
>> caller_contact and callee_contact values (from fix_nated_contact() above),
>> so when the requests pass through the sequential handling section of the
>> script and topology_hiding_match() does its fixups, the request URI domain
>> of the relayed request has the received IP:port values of the target UA
>> rather than the private IP:port values the UA provided during the initial
>> request that established the dialog.
>>
>> I can't wrap my head around how to fix this.  The initial requests work
>> because lookup() has the intelligence to distinguish the UAC's Contact from
>> the received IP:port at REGISTER-time, but I can't see how to achieve this
>> at dialog-creation time so sequential requests have the right RURI domain.
>> Force the caller_contact and callee_contact to the private values somehow,
>> and manage the route_set to point to the appropriate received IP:port?  I'm
>> not sure how to configure that if it is the solution.
>>
>> Any direction would be appreciated!
>>
>>
>> Regards,
>> Jeff
>>
>> _______________________________________________
>> 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 ahmedmunir007 at gmail.com  Tue Jan 26 17:06:38 2021
From: ahmedmunir007 at gmail.com (Ahmed Chohan)
Date: Tue, 26 Jan 2021 12:06:38 -0500
Subject: [OpenSIPS-Users] OPenSIPs 3.1 + CDRTool 9.9,
 Nodes IP Address and Call Direction for SIP Trace
Message-ID: <CAGMN=JeKmFzs7p-yP5gGjZmNo0w10KEyG4uC8HjHCqAZjLkzkQ@mail.gmail.com>

Hi,

I've recently installed and configured OpenSIPs 3.1 + CDRTool 9.9 on CentOS
7 platform. As per functionality based not facing any issues as able to get
SIP traces and accounting.

As per CDRTool 9.9 release, they've added feature nodes IP addresses along
with directions for  SIP trace. On OPenSIPs configuration, I've already
declared and configured tracer modules however, the issue I'm experiencing
is not showing call direction (INVITE, Trying, OK, etc) and nodes IP
address but only showing BYE header direction.

Please advise additional configuration I may be missing out as tried as
 followed tracer module document i.e. loaded module, modparameter and
configured in routing script along with CDRTool installation document.

-- 
Regards,

Ahmed Munir Chohan
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From chunyong.zhao at qq.com  Thu Jan 21 05:50:00 2021
From: chunyong.zhao at qq.com (=?ISO-8859-1?B?SmVmZnJleSBaaGFv?=)
Date: Thu, 21 Jan 2021 13:50:00 +0800
Subject: [OpenSIPS-Users] Question regarding Federated User Location Cluster
Message-ID: <tencent_AB9F4ECDF83591D301E79A56DE7F97BDC906@qq.com>

Dear Team
After reading through below&nbsp;Tutorial, I have some questions regarding&nbsp;Database and NoSQL setup model for Federated User Location Cluster.




https://opensips.org/Documentation/Tutorials-Distributed-User-Location-Federation


For example, for formal production system senario, two sites, with two opensips nodes for each site, HA mode for each site.
1. Should I deploy MySQL and Cassandra on each node? 4 MySQL instances and 4 Cassandra on each node?
2. Should I setup db replication among 4 MySQL instances or just standalone separated setup?
3. Should I put all 4&nbsp; Cassandra instances into one cluster?
4. For each HA virtual IP setup, what's the recommended tool, keepalived?




Thanks in advance.


Best wishes,
Jeffrey
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From krishdinesh at yahoo.com  Sun Jan 24 17:20:35 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Sun, 24 Jan 2021 17:20:35 +0000 (UTC)
Subject: [OpenSIPS-Users] REST API Challenge
References: <1990202090.2287782.1611508835989.ref@mail.yahoo.com>
Message-ID: <1990202090.2287782.1611508835989@mail.yahoo.com>

Hello,
I am trying to test a REST API call using the example code provided in the documentation. I am experiencing the following error and opensips would not start. I am running opensips 3.1.1 (x86_64/linux) on Debian 10. Any help would be appreciated. Curl is also installed and i am able to query a URL successfully Thank you
...# Example of querying a REST service to get the credit of an account$var(rc) = rest_get("https://getcredit.org/?account=$fU",                    $var(credit),                    $var(ct),                    $var(rcode));if ($var(rc) < 0) { xlog("rest_get() failed with $var(rc), acc=$fU\n"); send_reply(500, "Server Internal Error"); exit;}
if ($var(rcode) != 200) { xlog("L_INFO", "rest_get() rcode=$var(rcode), acc=$fU\n"); send_reply(403, "Forbidden"); exit;}


Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: INFO:rest_client:mod_init: Module initialized!Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: INFO:auth:mod_init: initializing...Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_cmd: Param [2] expected to be a variableJan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: Failed to fix command <rest_get>Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:fix_actions: fixing failed (code=-6) at /etc/opensips/opensips.cfg:232Jan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: CRITICAL:core:fix_expr: fix_actions errorJan 24 22:20:48 OpenSIPS /usr/sbin/opensips[739]: ERROR:core:main: failed to fix configuration with err code -6
Thank you,Dinesh
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From krishdinesh at yahoo.com  Thu Jan 28 17:33:16 2021
From: krishdinesh at yahoo.com (Dinesh Krishnamurthy)
Date: Thu, 28 Jan 2021 17:33:16 +0000 (UTC)
Subject: [OpenSIPS-Users] OpenSIPS - Asterisk issue
References: <394823969.166084.1611855196069.ref@mail.yahoo.com>
Message-ID: <394823969.166084.1611855196069@mail.yahoo.com>

Hi,
I am integrating OpenSIPS and Asterisk to use Asterisk to play media (typical media treatment)
I have a softphone registered to OpenSIPS and when i call a specific number, a simple prompt needs to be played from asterisk. I have the sip configuration and also extensions.conf file setup.
When i call the specific number, the SIP messages are exchanged but the call drops stating calling number not found (i have the number configured in asterisk though). In OpenSIPS.cfg all i am doing is just calling the function  sethostport("<Asterisk IP>:5060") when receiving the call at this number 
If i register the endpoints directly with Asterisk, i can hear the announcement as expected.
Not sure if i am missing something or is there anything that needs to be set specifically in OpenSIPS for this to work? 
Thank you,DK
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From spanda at 3clogic.com  Fri Jan 29 07:32:04 2021
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 29 Jan 2021 13:02:04 +0530
Subject: [OpenSIPS-Users] Need some help for auth_jwt module .
In-Reply-To: <CALf-Q46n528Cz40TXH7uSjcDGe+VJJeZ4JauwjZFjNF9GoTLYw@mail.gmail.com>
References: <CALf-Q46n528Cz40TXH7uSjcDGe+VJJeZ4JauwjZFjNF9GoTLYw@mail.gmail.com>
Message-ID: <CALf-Q45Jfv6Ldv7OMcA5LMD7dPgteNHP8-6z9n9uehXssKowaQ@mail.gmail.com>

Hi ,

Any help on this .

The public key which I am configuring in the secret column , opensips is
converting to that BLOB . Why so ?

*DBG:db_mysql:db_mysql_str2val: converting BLOB
[vP7wAX3vvI5zL97-7tIwRE2WJf3hTdCtgHNrzkpdlZW9bMEBdVsF3cePFANPrJkxiMqnCukzLqX3wjBbHaj6rZQtani23maeOCe9wvD6m1vImLW6tkne-5k8lbIRmifo9gdBiLQnG5J_rHZotJbRSNf2GTd5a1O7YpIb53wpBIWcuXU7DdlM-oMUZB6epSKjn2ujl413RyWBBp2WT1cvX9kKo6p2YXrWGF_zHNPJ37D7r5eccZXyEbwpWadI6XKVldhZF24RYX3vU7g1YuytgBY_eNJVwKjYgVoi12VrSo6MgaV7JicsX6d2pykFAmVM3To57O5rYdQHF92CpR7MJQ]*

As the public key is multiline text how will I put this in  the table ?




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


On Fri, Jan 22, 2021 at 1:44 PM Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi ,
> First time I am trying to use auth_jwt module with opensips 3.1 .
>
> When I am generating jwt token by using HS256 algorithm and adding the
> corresponding tag and key in opensips DB then opensips is able to decode
> the token successfully .
>
> When I am generating the token by using RS256 algorithm  and both private
> and public key in the corresponding to the token in the db then opensips is
> not able to decode the token using that key .
>
>  DBG:auth_jwt:jwt_authorize: Found 2 record for tag 1234567890
>  DBG:auth_jwt:jwt_authorize: Failed to decode jwt with DB secret
>  DBG:auth_jwt:jwt_authorize: Failed to decode jwt with DB secret
>
> for both the secrets this is the error I am getting . How will I do this ?
> What should I add in secret column corresponding to the token ?
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
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From johan at democon.be  Fri Jan 29 10:02:11 2021
From: johan at democon.be (johan)
Date: Fri, 29 Jan 2021 11:02:11 +0100
Subject: [OpenSIPS-Users] issue with reply_route , failure_route.
Message-ID: <010a3b0e-fab1-3604-84c8-a74563b0a787@democon.be>

Is it normal that $rs is only populated in on_reply route ?

Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
Onreply_route[handle_nat]: incoming reply rs [100]
Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
Onreply_route[handle_nat]: incoming reply rs [603]
Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
failure_route[missed_call] there is an error, we call rtpengine_delete
first, rs [<null>]

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From radcia at gmail.com  Fri Jan 29 11:30:31 2021
From: radcia at gmail.com (Rafael Domingos)
Date: Fri, 29 Jan 2021 08:30:31 -0300
Subject: [OpenSIPS-Users] Help for Create Proxy Sip witch OpenSips
Message-ID: <CAJ2qaJYrqhZG=S3ncJf1ngF6P6hyAxwcH0eg1POf=+bA_p5jug@mail.gmail.com>

HI,


My name is Rafael and i have a multiple PBXIP in my structure.
I would like redirect traffic (register, options, invites etc..)
based proxy sip. I would lik remove NAT configurations, today I need creat
rules of redirect ports and IP's for PBX.

I need this configuration:
I need only redirect sip connections for any asterisk. Exemplo: registre
UAC for domainA.com <http://domaina.com/> —->>> IP ASTeriskA domainB.com
<http://domainb.com/> —->>> IP ASTERISKB

Anybody can help me?


Att..
*Rafael Domingos /**  31-988485832*
*SKYPE:  radibraz at hotmail.com <radibraz at hotmail.com>*
*EMAIL:  radcia at gmail.com <radcia at gmail.com>*
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From david.villasmil.work at gmail.com  Fri Jan 29 11:58:32 2021
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 29 Jan 2021 11:58:32 +0000
Subject: [OpenSIPS-Users] Help for Create Proxy Sip witch OpenSips
In-Reply-To: <CAJ2qaJYrqhZG=S3ncJf1ngF6P6hyAxwcH0eg1POf=+bA_p5jug@mail.gmail.com>
References: <CAJ2qaJYrqhZG=S3ncJf1ngF6P6hyAxwcH0eg1POf=+bA_p5jug@mail.gmail.com>
Message-ID: <CAFGRPVphCPXuk8BrDk_b7F04KDy6b2vESYJ8a4_y33wj1NRToQ@mail.gmail.com>

Hello Rafael,

You might want to start by installing Opensips and going through the
quick-start:

https://www.opensips.org/Documentation/Tutorials-GettingStarted

Once you've done that, and gone through the configuration file, you can ask
more pointed questions.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337


On Fri, Jan 29, 2021 at 11:31 AM Rafael Domingos <radcia at gmail.com> wrote:

> HI,
>
>
> My name is Rafael and i have a multiple PBXIP in my structure.
> I would like redirect traffic (register, options, invites etc..)
> based proxy sip. I would lik remove NAT configurations, today I need creat
> rules of redirect ports and IP's for PBX.
>
> I need this configuration:
> I need only redirect sip connections for any asterisk. Exemplo: registre
> UAC for domainA.com <http://domaina.com/> —->>> IP ASTeriskA domainB.com
> <http://domainb.com/> —->>> IP ASTERISKB
>
> Anybody can help me?
>
>
> Att..
> *Rafael Domingos /**  31-988485832*
> *SKYPE:  radibraz at hotmail.com <radibraz at hotmail.com>*
> *EMAIL:  radcia at gmail.com <radcia at gmail.com>*
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From john at voxtelesys.net  Fri Jan 29 14:13:23 2021
From: john at voxtelesys.net (John Burke)
Date: Fri, 29 Jan 2021 08:13:23 -0600
Subject: [OpenSIPS-Users] issue with reply_route , failure_route.
In-Reply-To: <010a3b0e-fab1-3604-84c8-a74563b0a787@democon.be>
References: <010a3b0e-fab1-3604-84c8-a74563b0a787@democon.be>
Message-ID: <7de179848c5feb9053ca338beea84e71@voxtelesys.net>


Hey Johan,

Failure routes are in the context of request [1] - from the docs: "Note that inside the 'failure_route', the request that initiated the transaction is being processed, and not its reply."

You can access reply context from failure route by specifying the context of $rs:

$(<context>name(subname)[index]{transformation})
$(<reply>rs)

[1] https://www.opensips.org/Documentation/Script-Routes-3-1#toc3

Thanks,
John
----- Original Message -----
From: johan (johan at democon.be)
Date: 01/29/21 04:04
To: OpenSIPS users mailling list (users at lists.opensips.org)
Subject: [OpenSIPS-Users] issue with reply_route , failure_route.

Is it normal that $rs is only populated in on_reply route ?

Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
Onreply_route[handle_nat]: incoming reply rs [100]
Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
Onreply_route[handle_nat]: incoming reply rs [603]
Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
failure_route[missed_call] there is an error, we call rtpengine_delete
first, rs [<null>]


_______________________________________________
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  Sat Jan 30 11:10:11 2021
From: johan at democon.be (johan)
Date: Sat, 30 Jan 2021 12:10:11 +0100
Subject: [OpenSIPS-Users] issue with reply_route , failure_route.
In-Reply-To: <7de179848c5feb9053ca338beea84e71@voxtelesys.net>
References: <010a3b0e-fab1-3604-84c8-a74563b0a787@democon.be>
 <7de179848c5feb9053ca338beea84e71@voxtelesys.net>
Message-ID: <ad233865-3c12-0c5c-573b-ac23d0929aa7@democon.be>

for your info : I caught the 603 reply in onreply_route, putted an avp
to true. 

Then using that avp in failure_route, I rewrote the reply to 100 trying
and issued a reinvite.


On 29/01/2021 15:13, John Burke via Users wrote:
> Hey Johan,
>
> Failure routes are in the context of request [1] - from the docs:
> "Note that inside the 'failure_route', the request that initiated the
> transaction is being processed, and not its reply."
>
> You can access reply context from failure route by specifying the
> context of $rs:
>
> $(/<context>/*name*/(subname)[index]{transformation}/)
> $(<reply>rs)
>
> [1] https://www.opensips.org/Documentation/Script-Routes-3-1#toc3
>
> Thanks,
> John
>
> ----- Original Message -----
> ------------------------------------------------------------------------
> From: johan (johan at democon.be <mailto:johan at democon.be>)
> Date: 01/29/21 04:04
> To: OpenSIPS users mailling list (users at lists.opensips.org
> <mailto:users at lists.opensips.org>)
> Subject: [OpenSIPS-Users] issue with reply_route , failure_route.
>
> Is it normal that $rs is only populated in on_reply route ?
>
> Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
> callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
> Onreply_route[handle_nat]: incoming reply rs [100]
> Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
> callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
> Onreply_route[handle_nat]: incoming reply rs [603]
> Jan 29 04:57:07 SBC-01 /data/opensips/sbin/opensips[9884]:
> callid=4dcae6e2542525c715d2890c178a9968 at 176.241.248.19:
> failure_route[missed_call] there is an error, we call rtpengine_delete
> first, rs [<null>]
>
> ------------------------------------------------------------------------
> _______________________________________________
> 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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