From prathibhab.tvm at gmail.com  Fri Sep  1 03:50:26 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 09:20:26 +0530
Subject: [OpenSIPS-Users] SIP registration failed
Message-ID: <CAJh_-cB8xMtASvgo=CnMpNsoM8nHxQqHUoeEHWxtSVuZBuODZQ@mail.gmail.com>

I've created two accounts: 1001 at bp3.erss.in and 1002 at bp3.erss.in in the
opensips-cp application. When I tried to register it in soft phone it shows
registration failed.

My configuration file is attached.

Expecting your help to resolve this issue.
-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Fri Sep  1 08:14:05 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 13:44:05 +0530
Subject: [OpenSIPS-Users] Error in opensips.cfg file
In-Reply-To: <CAMVLSOAZiYUdPx0Q5KsJrqyesqBksRJ4tWUA8dmyHC3_=mdLgw@mail.gmail.com>
References: <CAJh_-cAjdbNf_yv9Zd2KxCqpkX6o4qiMem=tGLd7SRMOkBUoSw@mail.gmail.com>
 <CAJh_-cCUO5Jg72x0Q9BzKpXgJQdC2hizUSfzpqCq5MY=GepCBg@mail.gmail.com>
 <CAJh_-cBnr8pKR9cCjzpKDD+xgZoFf6od0jzc80nojvc+2jOenA@mail.gmail.com>
 <CAJh_-cCFLG=Wvw8e8iM1jD2W55Y+9x7gkjnCLPQi-VtOjr6kBQ@mail.gmail.com>
 <CAMVLSOAZiYUdPx0Q5KsJrqyesqBksRJ4tWUA8dmyHC3_=mdLgw@mail.gmail.com>
Message-ID: <CAJh_-cCR4rpVZu5ELQih8EwKLvfr4vry7wjPWC3X6uXmB1DaOg@mail.gmail.com>

This issue is resolved after migration to the latest version.

On Fri, 1 Sept 2023 at 13:41, Alejandro Mejia Evertsz <
amejia at voxdatacomm.com> wrote:

> Hello Phrathib
>
> It seems you're using some commands used in version 2.
> You may need to look for the correct syntax to use them on version 3.
> There is a document on the website that explains migrating from 2 to 3.
> https://www.opensips.org/Documentation/Migration-2-4-0-to-3-0-0
>
> Hope it helps
>
> On Wed, Aug 30, 2023, 11:17 PM Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> Aug 31 04:14:49 [339966] CRITICAL:Traceback (last included file at the
>> bottom):
>> Aug 31 04:14:49 [339966] CRITICAL: 0. //etc/opensips/opensips.cfg
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:12-14: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:
>> Aug 31 04:14:49 [339966] CRITICAL:      # check if the clients are using
>> WebSockets or WebSocketSecure
>> Aug 31 04:14:49 [339966] CRITICAL:      if (proto == "WS" || proto ==
>> "WSS")
>> Aug 31 04:14:49 [339966] CRITICAL:      ^~~
>> Aug 31 04:14:49 [339966] CRITICAL:              setflag(SRC_WS);
>> Aug 31 04:14:49 [339966] CRITICAL:
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:12-14: bare word <proto> found, command
>> calls need '()'
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:12-14: bad command!)
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:16-17: bad command!)
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:18-20: bad command!)
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:27-29: bare word <proto> found, command
>> calls need '()'
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:27-29: bad command: missing ';'?
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:31-32: bad command!)
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:183:32-33: bad command!)
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:184:17-18: bad arguments for command <setflag>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:193:17-23: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:193:23-24: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:194:13-19: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:194:19-20: bad arguments for command <setbflag>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:239:16-22: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:239:22-23: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:239:38-44: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:239:44-45: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:241:21-27: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:241:27-28: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:241:44-50: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:241:50-51: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:243:22-28: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:243:28-29: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:243:44-50: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:243:50-51: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:245:22-28: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:245:28-29: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:245:45-51: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:245:51-52: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:257:16-22: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:257:22-23: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:257:38-44: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:257:44-45: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:259:21-27: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:259:27-28: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:259:44-50: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:259:50-51: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:261:22-28: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:261:28-29: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:261:44-50: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:261:50-51: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:263:22-28: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:263:28-29: bad arguments for command <isflagset>
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:263:45-51: syntax error
>> Aug 31 04:14:49 [339966] CRITICAL:core:yyerror: parse error in
>> //etc/opensips/opensips.cfg:263:51-52: bad arguments for command
>> <isbflagset>
>> Aug 31 04:14:49 [339966] ERROR:core:parse_opensips_cfg: bad config file
>> (46 errors)
>> Aug 31 04:14:49 [339966] ERROR:core:main: failed to parse config file
>> (null)
>> Aug 31 04:14:49 [339966] NOTICE:core:main: Exiting....
>>
>> On Thu, 31 Aug 2023 at 10:04, Prathibha B <prathibhab.tvm at gmail.com>
>> wrote:
>>
>>> bare word <proto> found, command calls need '()'
>>>
>>> On Mon, 21 Aug 2023 at 13:34, Prathibha B <prathibhab.tvm at gmail.com>
>>> wrote:
>>>
>>>> if(proto === WSS)
>>>>
>>>> On Mon, 21 Aug 2023 at 13:33, Prathibha B <prathibhab.tvm at gmail.com>
>>>> wrote:
>>>>
>>>>> Aug 21 08:02:24 [210480] CRITICAL:Traceback (last included file at the
>>>>> bottom):
>>>>> Aug 21 08:02:24 [210480] CRITICAL: 0. //etc/opensips/opensips.cfg
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:16-18: syntax error
>>>>> Aug 21 08:02:24 [210480] CRITICAL:      if (is_method("REGISTER"))
>>>>> Aug 21 08:02:24 [210480] CRITICAL:      {
>>>>> Aug 21 08:02:24 [210480] CRITICAL:              if(proto_wss == WSS) {
>>>>> Aug 21 08:02:24 [210480] CRITICAL:              ^~~
>>>>> Aug 21 08:02:24 [210480] CRITICAL:
>>>>>  fix_nated_register();
>>>>> Aug 21 08:02:24 [210480] CRITICAL:              }
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:16-18: bare word <proto_wss> found, command
>>>>> calls need '()'
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:16-18: bad command!)
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:22-23: bare word <WSS> found, command calls
>>>>> need '()'
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:22-23: bad command: missing ';'?
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:250:24-25: bad command!)
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:256:3-5: syntax error
>>>>> Aug 21 08:02:24 [210480] CRITICAL:core:yyerror: parse error in
>>>>> //etc/opensips/opensips.cfg:256:3-5:
>>>>> Aug 21 08:02:24 [210480] ERROR:core:parse_opensips_cfg: bad config
>>>>> file (8 errors)
>>>>> Aug 21 08:02:24 [210480] ERROR:core:main: failed to parse config file
>>>>> (null)
>>>>> Aug 21 08:02:24 [210480] NOTICE:core:main: Exiting....
>>>>>
>>>>> --
>>>>> Regards,
>>>>> B.Prathibha
>>>>>
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> B.Prathibha
>>>>
>>>
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>> _______________________________________________
>> 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,
B.Prathibha
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From prathibhab.tvm at gmail.com  Fri Sep  1 08:32:31 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 14:02:31 +0530
Subject: [OpenSIPS-Users] systemctl status opensips
In-Reply-To: <CAJh_-cBrGTW8BxaM0MNutoDvrYjV=VOS=e7P2hHY9PqUYv6eZg@mail.gmail.com>
References: <CAJh_-cBrGTW8BxaM0MNutoDvrYjV=VOS=e7P2hHY9PqUYv6eZg@mail.gmail.com>
Message-ID: <CAJh_-cAFwoUfnGN3Zx_B3u1Y4zTit0UZpCgr5-o+R86RZv-ssQ@mail.gmail.com>

I'm getting  *Active: active (exited). How to resolve this issue?*

On Thu, 31 Aug 2023 at 14:22, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> opensips.service - LSB: Start the OpenSIPS SIP server
>      Loaded: loaded (/etc/init.d/opensips; generated)
>    *  Active: active (exited) since Thu 2023-08-31 08:17:20 UTC; 32min
> ago*
>        Docs: man:systemd-sysv-generator(8)
>     Process: 347316 ExecStart=/etc/init.d/opensips start (*code=exited,
> status=0/SUCCESS*)
>         CPU: 44ms
>
> Aug 31 08:17:20 ip-172-31-34-24 systemd[1]: Starting LSB: Start the
> OpenSIPS SIP server...
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347316]:  * Starting opensips
> opensips
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]: Listening on
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]:              udp:
> 127.0.0.1 [127.0.0.1]:5060
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]:              tls:
> 127.0.0.1 [127.0.0.1]:5061
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]:              ws:
> 127.0.0.1 [127.0.0.1]:8080
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]:              wss:
> 127.0.0.1 [127.0.0.1]:443
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347333]: Aliases:
> Aug 31 08:17:20 ip-172-31-34-24 opensips[347316]:  already running
> ...done.
> Aug 31 08:17:20 ip-172-31-34-24 systemd[1]: Started LSB: Start the
> OpenSIPS SIP server.
>
> *netstat -ltnup doesn't show 5060, 5061, 8080 and 443*
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Fri Sep  1 09:59:52 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 15:29:52 +0530
Subject: [OpenSIPS-Users] opensips not getting started
Message-ID: <CAJh_-cD_K1xPtXicnU0fj36GDe306KfLcYRCT_4L_QNfKvaUmA@mail.gmail.com>

while running

systemctl start opensips
systemctl status opensips

opensips.service - LSB: Start the OpenSIPS SIP server
     Loaded: loaded (/etc/init.d/opensips; generated)
     Active: active (exited) since Fri 2023-09-01 09:10:43 UTC; 47min ago
       Docs: man:systemd-sysv-generator(8)
    Process: 361274 ExecStart=/etc/init.d/opensips start (code=exited,
status=0/SUCCESS)
        CPU: 45ms

Sep 01 09:10:43 ip-172-31-34-24 systemd[1]: Starting LSB: Start the
OpenSIPS SIP server...
Sep 01 09:10:43 ip-172-31-34-24 opensips[361274]:  * Starting opensips
opensips
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]: Listening on
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              udp:
172.31.34.24 [172.31.34.24]:5060
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              tls:
172.31.34.24 [172.31.34.24]:5061
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              ws:
172.31.34.24 [172.31.34.24]:8080
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              wss:
172.31.34.24 [172.31.34.24]:443
Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]: Aliases:
Sep 01 09:10:43 ip-172-31-34-24 opensips[361274]:  already running
...done.
Sep 01 09:10:43 ip-172-31-34-24 systemd[1]: Started LSB: Start the OpenSIPS
SIP server.

the ports don't appear in nestat -ltnp

The configuration file is also attached. How to resolve this?
-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Fri Sep  1 10:11:59 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 15:41:59 +0530
Subject: [OpenSIPS-Users] opensips not getting started
In-Reply-To: <CAJh_-cD_K1xPtXicnU0fj36GDe306KfLcYRCT_4L_QNfKvaUmA@mail.gmail.com>
References: <CAJh_-cD_K1xPtXicnU0fj36GDe306KfLcYRCT_4L_QNfKvaUmA@mail.gmail.com>
Message-ID: <CAJh_-cDyzcsq6_Y6U_R3oNV357tSwNWn8woWtV91SYu_Dpgs_Q@mail.gmail.com>

● opensips.service - LSB: Start the OpenSIPS SIP server
     Loaded: loaded (/etc/init.d/opensips; generated)
     Active: active (exited) since Fri 2023-09-01 10:10:10 UTC; 7s ago
       Docs: man:systemd-sysv-generator(8)
    Process: 361928 ExecStart=/etc/init.d/opensips start (code=exited,
status=0/SUCCESS)
        CPU: 44ms

Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
NOTICE:core:main: using 128 MB of shared memory, allocator: Q_MALLOC_DBG
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
NOTICE:core:main: using 4 MB of private process memory, allocator:
Q_MALLOC_DBG
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
NOTICE:signaling:mod_init: initializing module ...
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
WARNING:usrloc:ul_check_config: 'db_mode' is now deprecated, use
'working_mode_preset'!
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
ERROR:tls_mgm:load_tls_library: No TLS library module loaded
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
ERROR:core:init_mod: failed to initialize module tls_mgm
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
ERROR:core:main: failed to initialize modules!
Sep 01 10:10:10 ip-172-31-34-24 opensips[361947]: Sep  1 10:10:10 [361947]
NOTICE:core:main: Exiting....
Sep 01 10:10:10 ip-172-31-34-24 opensips[361928]:  already running
...done.
Sep 01 10:10:10 ip-172-31-34-24 systemd[1]: Started LSB: Start the OpenSIPS
SIP server.


*I've loaded tls_mgm.so in /lib64/opensips/modules. But still it is showing
the error.*

On Fri, 1 Sept 2023 at 15:29, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> while running
>
> systemctl start opensips
> systemctl status opensips
>
> opensips.service - LSB: Start the OpenSIPS SIP server
>      Loaded: loaded (/etc/init.d/opensips; generated)
>      Active: active (exited) since Fri 2023-09-01 09:10:43 UTC; 47min ago
>        Docs: man:systemd-sysv-generator(8)
>     Process: 361274 ExecStart=/etc/init.d/opensips start (code=exited,
> status=0/SUCCESS)
>         CPU: 45ms
>
> Sep 01 09:10:43 ip-172-31-34-24 systemd[1]: Starting LSB: Start the
> OpenSIPS SIP server...
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361274]:  * Starting opensips
> opensips
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]: Listening on
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              udp:
> 172.31.34.24 [172.31.34.24]:5060
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              tls:
> 172.31.34.24 [172.31.34.24]:5061
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              ws:
> 172.31.34.24 [172.31.34.24]:8080
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]:              wss:
> 172.31.34.24 [172.31.34.24]:443
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361292]: Aliases:
> Sep 01 09:10:43 ip-172-31-34-24 opensips[361274]:  already running
> ...done.
> Sep 01 09:10:43 ip-172-31-34-24 systemd[1]: Started LSB: Start the
> OpenSIPS SIP server.
>
> the ports don't appear in nestat -ltnp
>
> The configuration file is also attached. How to resolve this?
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Fri Sep  1 12:10:50 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 1 Sep 2023 17:40:50 +0530
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS library
	module loaded
Message-ID: <CAJh_-cBoMSqn7w5X0dpWAdd=vaww2UgJ06qg-KKzBvxHs55Y2Q@mail.gmail.com>

I've loaded tls_mgm module in /lib64/opensips/modules. Still getting this
error.

NOTICE:core:main: version: opensips 3.5.0-dev (x86_64/linux)
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
NOTICE:core:main: using 128 MB of shared memory, allocator: Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
NOTICE:core:main: using 4 MB of private process memory, allocator:
Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
NOTICE:signaling:mod_init: initializing module ...
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
ERROR:tls_mgm:load_tls_library: No TLS library module loaded
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
ERROR:core:init_mod: failed to initialize module tls_mgm
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
ERROR:core:main: failed to initialize modules!
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
NOTICE:core:main: Exiting...

-- 
Regards,
B.Prathibha
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From alberto.rinaudo at gmail.com  Fri Sep  1 13:43:06 2023
From: alberto.rinaudo at gmail.com (Alberto)
Date: Fri, 1 Sep 2023 14:43:06 +0100
Subject: [OpenSIPS-Users] rest_post logs
Message-ID: <CANoA6_+pO=UJwvMcbqop-_-eUBO5UkyyNXHp8LdNNuq8_j8TTA@mail.gmail.com>

Hi,

Is there a way to turn off logs for rest_post requests?

This is my log settings:
log_level=-1
xlog_level=-1
log_stdout=yes
log_stderror=yes
log_facility=LOG_LOCAL0

This is the type of output I want to suppress:
Sep 01 13:31:48 opensips opensips[59740]: > POST /api/opensips/doit HTTP/1.1
Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
Sep 01 13:31:48 opensips opensips[59740]: Content-Type: application/json
Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
Sep 01 13:31:48 opensips opensips[59740]:
Sep 01 13:31:48 opensips opensips[59740]: * upload completely sent off: 156
out of 156 bytes
Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not supporting
multiuse
Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep 2023 13:31:48
GMT
Sep 01 13:31:48 opensips opensips[59740]: < Content-Type: application/json;
charset=utf-8
Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
Sep 01 13:31:48 opensips opensips[59740]: < Access-Control-Allow-Origin: *
Sep 01 13:31:48 opensips opensips[59740]: < Content-Security-Policy:
default-src 'self';base-uri 'self';font-src 'self' https: data:;form-action
'self';frame-ancestors 'self';img-src 'self' data:;object-src
'none';script-src 'self';script-src-attr 'none';style-src 'self' https:
'unsafe-inline';upgrade-insecure-requests
Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Opener-Policy:
same-origin
Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Resource-Policy:
same-origin
Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy: no-referrer
Sep 01 13:31:48 opensips opensips[59740]: < Strict-Transport-Security:
max-age=15552000; includeSubDomains
Sep 01 13:31:48 opensips opensips[59740]: < X-Content-Type-Options: nosniff
Sep 01 13:31:48 opensips opensips[59740]: < X-DNS-Prefetch-Control: off
Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options: noopen
Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options: SAMEORIGIN
Sep 01 13:31:48 opensips opensips[59740]: <
X-Permitted-Cross-Domain-Policies: none
Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
Sep 01 13:31:48 opensips opensips[59740]: <
Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host 127.0.0.1
left intact

Thanks
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From amejia at voxdatacomm.com  Fri Sep  1 15:41:10 2023
From: amejia at voxdatacomm.com (Alejandro Mejia Evertsz)
Date: Fri, 1 Sep 2023 09:41:10 -0600
Subject: [OpenSIPS-Users] mid_registrar and nathelper problem
Message-ID: <CAMVLSOARUfVxhZ3ER0jffJGpgDOz4wg-z_hAUVwFo53Vx-i58g@mail.gmail.com>

Hi!
This is my first OpenSIPS installation, and I'm trying to achieve a simple
sip proxy for extensions to register to a PBX.
Got some configurations from samples all over docs and google searches and
everything seems to work fine until there's a call from the PBX to a
registered endpoint.
The endpoint rings, but when it answers the call, it sends OK to proxy, and
proxy to PBX, but when PBX sends ACK and proxy to endpoint, the ACK gets
lost on the proxy (never reaching the endpoint).

Same problem reported by another user on Dec. 2020 here:
https://opensips.org/pipermail/users/2020-December/044146.html
He didn't got any replies to his thread, so I would appreciate anyone's
help here.

Configuration file:
https://pastebin.com/GaQEMzmb

SIP Trace:
https://pastebin.com/3cREhszJ

As you can see, the ACK doesn't even try to leave the public (WAN)
interface of the proxy.

Thanks in advance for your help.

Alejandro
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From ray at hero.co.nz  Sat Sep  2 00:17:49 2023
From: ray at hero.co.nz (Ray Jackson)
Date: Sat, 2 Sep 2023 12:17:49 +1200
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS registrations
Message-ID: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>

Hi all,

I'm facing a weird issue which I think is related to broken TCP socket 
reuse logic where the wrong client is receiving incoming calls due to 
the wrong socket being used for the incoming INVITE.

The scenario is when I have 2 clients registering using TLS behind NAT 
at the same Public IPv4 address and both clients are using the same 
private port number.  So client 1 registers and the Via and contact 
header looks like:

Via: SIP/2.0/TLS 
192.168.42.162:5062;branch=z9hG4bK1409895926;rport;alias Contact: 
<sip:201 at 192.168.42.162:5062;transport=tls>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4>"

Client 2 registers from behind the same Public IPv4 address and the Via 
and contact header looks like:

Via: SIP/2.0/TLS 192.168.42.186:5062;branch=z9hG4bK-aff1f3b3 Contact: 
<sip:202 at 192.168.42.186:5062;transport=tls>;expires=300

The location table shows Client 1 received field of 103.212.1.2:5062 and 
Client 103.212.1.2:23456

When a call comes in for Client 1 the location lookup seems to return 
the correct 'received' address and port (e.g. 103.212.1.2:5062) and all 
the logs indicate that this is where the SIP INVITE *should* be going to 
(in the $du field).  However when you check the SIP traffic it selects 
Client 2's socket and the traffic goes to port 23456 instead of 5062.

I think this is related somehow to the TCP port reuse logic inside 
Opensips.  My suspicion is that Opensips is looking at the Contact or 
Via port number (which is the same for both client 1 and 2) and then 
somehow mapping this to the wrong TCP received socket.

Does anybody have any suggestions here?  Should I be fixing the NAT in 
the Contact header (using fix_nated_contact).  I read somewhere that you 
shouldn't rewrite the Contact header to avoid problems with sending a 
different Contact URI to the client on calls.  Or is this issue more 
related to the Via header and the TCP port reuse logic looking at this 
port instead of the actual received port when choosing the outgoing socket?

FYI: I am using both force_rport() and fix_nated_register() for incoming 
registrations from these clients and matching_mode of 0 in usrloc.  
However, I am not using fix_nated_contact() for registrations.

Thanks,

Ray

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From prathibhab.tvm at gmail.com  Sat Sep  2 09:56:43 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 2 Sep 2023 15:26:43 +0530
Subject: [OpenSIPS-Users] opensips startup errors
Message-ID: <CAJh_-cBAhKWQLyYj3E0mRhRE9nc0kxUOMKhcs3WXG4TgsC=mUg@mail.gmail.com>

ERROR:rtpengine:send_rtpe_command: can't send (#7 iovec buffers) command to
a RTP proxy (111:Connection refused)
Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:47:27 [147483] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:47:30 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:47:30 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:47:30 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:47:50 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:47:50 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:47:50 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:48:09 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:48:09 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:48:09 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:48:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:48:29 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:48:29 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:48:29 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:48:29 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:48:29 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:48:29 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:48:48 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:48:48 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:48:48 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:49:07 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:49:07 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:49:07 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:49:26 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:49:26 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:49:26 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:49:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:49:46 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:49:46 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:49:46 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:49:46 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:49:46 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:49:46 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
^C
root at ip-172-31-34-24:/etc/opensips# KISep  2 09:50:05 [147490]
ERROR:proto_wss:wss_conn_init: no TLS server domain found
Sep  2 09:50:05 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:50:05 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
LL^C
root at ip-172-31-34-24:/etc/opensips# vi /usr/local/etc/opensips/opensips.cfg
root at ip-172-31-34-24:/etc/opensips# Sep  2 09:50:44 [147490]
ERROR:proto_wss:wss_conn_init: no TLS server domain found
Sep  2 09:50:44 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:50:44 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:51:04 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:51:04 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:51:04 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:51:04 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:51:04 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:51:04 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:51:22 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:51:22 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:51:22 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:51:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:51:40 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:51:40 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:51:40 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:52:00 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:52:00 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:52:00 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:52:19 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:52:19 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:52:19 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:52:19 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:52:19 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:52:19 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:52:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:52:39 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:52:39 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:52:39 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:52:57 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:52:57 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:52:57 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:53:16 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:53:16 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:53:16 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:53:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:53:36 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:53:36 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:53:36 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:53:36 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:53:36 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:53:36 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:53:55 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:53:55 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:53:55 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:54:14 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:54:14 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:54:14 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:54:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:54:34 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:54:34 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:54:34 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:54:53 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:54:53 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:54:53 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:54:53 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:54:53 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:54:53 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:55:12 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:55:12 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:55:12 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
iovec buffers) command to a RTP proxy (111:Connection refused)
Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout waiting
reply from a RTP proxy
Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
127.0.0.0:60000> does not respond, disable it
Sep  2 09:55:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
to ping
Sep  2 09:55:32 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:55:32 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:55:32 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct
Sep  2 09:55:52 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
domain found
Sep  2 09:55:52 [147490] ERROR:core:handle_io: failed to do proto 6
specific init for conn 0x7f2c4123d130
Sep  2 09:55:52 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
the tls_domain pointer in the SSL struct

-- 
Regards,
B.Prathibha
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From rvg at rvgeerligs.nl  Sat Sep  2 11:28:53 2023
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Sat, 02 Sep 2023 11:28:53 +0000
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS
	registrations
In-Reply-To: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
References: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
Message-ID: <8ca4edac68dae40a21cbca67aff54c836f295ee7@rvgeerligs.nl>

Hi Ray,

I am interested the solution or comments you receive. I have a similar problem. I have no voice or oneway voice on one of the clients. But I could work around it as I am using softphones on iphones as client. My work around was to turn wifi off on one of the clients, so this client gets the public IP of the phone subscription. But I still have this problem is both clients are in the same NAT.

Regards,

Ronald



September 2, 2023 at 2:17 AM, "Ray Jackson" <ray at hero.co.nz> wrote:


> 
> Hi all,
> 
>  
> 
> I'm facing a weird issue which I think is related to broken TCP socket reuse logic where the wrong client is receiving incoming calls due to the wrong socket being used for the incoming INVITE. 
> 
>  
> 
> The scenario is when I have 2 clients registering using TLS behind NAT at the same Public IPv4 address and both clients are using the same private port number.  So client 1 registers and the Via and contact header looks like: 
> 
>  
> 
> **Via: ** SIP/2.0/TLS 192.168.42.162:5062;**branch=**z9hG4bK1409895926;rport;alias**Contact: ** <sip:201 at 192.168.42.162:5062;**transport=**tls>;**reg-id=**2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4> urn:uuid:00000000-0000-1000-8000-C074AD928AC4 "
> 
>  
> 
> Client 2 registers from behind the same Public IPv4 address and the Via and contact header looks like:
> 
>  
> 
> **Via: ** SIP/2.0/TLS 192.168.42.186:5062;**branch=**z9hG4bK-aff1f3b3**Contact: ** <sip:202 at 192.168.42.186:5062;**transport=**tls>;expire**s=**300
> 
>  
> 
> The location table shows Client 1 received field of 103.212.1.2:5062 and Client 103.212.1.2:23456 
> 
>  
> 
> When a call comes in for Client 1 the location lookup seems to return the correct 'received' address and port (e.g. 103.212.1.2:5062) and all the logs indicate that this is where the SIP INVITE *should* be going to (in the $du field).  However when you check the SIP traffic it selects Client 2's socket and the traffic goes to port 23456 instead of 5062.
> 
>  
> 
> I think this is related somehow to the TCP port reuse logic inside Opensips.  My suspicion is that Opensips is looking at the Contact or Via port number (which is the same for both client 1 and 2) and then somehow mapping this to the wrong TCP received socket.
> 
>  
> 
> Does anybody have any suggestions here?  Should I be fixing the NAT in the Contact header (using fix_nated_contact).  I read somewhere that you shouldn't rewrite the Contact header to avoid problems with sending a different Contact URI to the client on calls.  Or is this issue more related to the Via header and the TCP port reuse logic looking at this port instead of the actual received port when choosing the outgoing socket?
> 
>  
> 
> FYI: I am using both force_rport() and fix_nated_register() for incoming registrations from these clients and matching_mode of 0 in usrloc.  However, I am not using fix_nated_contact() for registrations.
> 
>  
> 
> Thanks,
> 
>  
> 
> Ray
>
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From rvg at rvgeerligs.nl  Sat Sep  2 15:31:45 2023
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Sat, 02 Sep 2023 15:31:45 +0000
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS
	registrations
In-Reply-To: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
References: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
Message-ID: <582654f0c7b35d24244c77b4185fb3758fec6000@rvgeerligs.nl>

Hi, again,

Did this work any time before?
I have force_rport(),  fix_nated_register() active and as follows:



if (nat_uac_test("diff-ip-src-contact")) {



if (is_method("REGISTER")) {



fix_nated_register();



setbflag("NAT");



} else {



fix_nated_contact();  <<<<<<<<<



setflag("NAT");



}



}


But in my tcpdump I only see 1 natted address. Which would mean opensips suddenly forgets the register with the other natted address  fix_nated_register().
we would like to see both natted ip adresses. The public IPv4 address is the only one addressed.


Regards

Ronald






September 2, 2023 at 2:17 AM, "Ray Jackson" <ray at hero.co.nz> wrote:


> 
> Hi all,
> 
>  
> 
> I'm facing a weird issue which I think is related to broken TCP socket reuse logic where the wrong client is receiving incoming calls due to the wrong socket being used for the incoming INVITE. 
> 
>  
> 
> The scenario is when I have 2 clients registering using TLS behind NAT at the same Public IPv4 address and both clients are using the same private port number.  So client 1 registers and the Via and contact header looks like: 
> 
>  
> 
> **Via: ** SIP/2.0/TLS 192.168.42.162:5062;**branch=**z9hG4bK1409895926;rport;alias**Contact: ** <sip:201 at 192.168.42.162:5062;**transport=**tls>;**reg-id=**2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4> urn:uuid:00000000-0000-1000-8000-C074AD928AC4 "
> 
>  
> 
> Client 2 registers from behind the same Public IPv4 address and the Via and contact header looks like:
> 
>  
> 
> **Via: ** SIP/2.0/TLS 192.168.42.186:5062;**branch=**z9hG4bK-aff1f3b3**Contact: ** <sip:202 at 192.168.42.186:5062;**transport=**tls>;expire**s=**300
> 
>  
> 
> The location table shows Client 1 received field of 103.212.1.2:5062 and Client 103.212.1.2:23456 
> 
>  
> 
> When a call comes in for Client 1 the location lookup seems to return the correct 'received' address and port (e.g. 103.212.1.2:5062) and all the logs indicate that this is where the SIP INVITE *should* be going to (in the $du field).  However when you check the SIP traffic it selects Client 2's socket and the traffic goes to port 23456 instead of 5062.
> 
>  
> 
> I think this is related somehow to the TCP port reuse logic inside Opensips.  My suspicion is that Opensips is looking at the Contact or Via port number (which is the same for both client 1 and 2) and then somehow mapping this to the wrong TCP received socket.
> 
>  
> 
> Does anybody have any suggestions here?  Should I be fixing the NAT in the Contact header (using fix_nated_contact).  I read somewhere that you shouldn't rewrite the Contact header to avoid problems with sending a different Contact URI to the client on calls.  Or is this issue more related to the Via header and the TCP port reuse logic looking at this port instead of the actual received port when choosing the outgoing socket?
> 
>  
> 
> FYI: I am using both force_rport() and fix_nated_register() for incoming registrations from these clients and matching_mode of 0 in usrloc.  However, I am not using fix_nated_contact() for registrations.
> 
>  
> 
> Thanks,
> 
>  
> 
> Ray
>
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From prathibhab.tvm at gmail.com  Sun Sep  3 09:04:20 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sun, 3 Sep 2023 14:34:20 +0530
Subject: [OpenSIPS-Users] opensips startup errors
In-Reply-To: <CAJh_-cBAhKWQLyYj3E0mRhRE9nc0kxUOMKhcs3WXG4TgsC=mUg@mail.gmail.com>
References: <CAJh_-cBAhKWQLyYj3E0mRhRE9nc0kxUOMKhcs3WXG4TgsC=mUg@mail.gmail.com>
Message-ID: <CAJh_-cBP4T_zFnNH2p8u0ptZ1sNiSGFKyv2FP2U4s9eqb2ZWPg@mail.gmail.com>

getting the above errors while checking opensips status using the command:

systemctl status opensips

On Sat, 2 Sept 2023 at 15:26, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> ERROR:rtpengine:send_rtpe_command: can't send (#7 iovec buffers) command
> to a RTP proxy (111:Connection refused)
> Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:47:27 [147483] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:47:27 [147483] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:47:30 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:47:30 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:47:30 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:47:50 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:47:50 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:47:50 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:48:09 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:48:09 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:48:09 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:48:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:48:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:48:29 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:48:29 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:48:29 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:48:29 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:48:29 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:48:29 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:48:48 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:48:48 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:48:48 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:49:07 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:49:07 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:49:07 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:49:26 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:49:26 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:49:26 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:49:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:49:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:49:46 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:49:46 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:49:46 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:49:46 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:49:46 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:49:46 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> ^C
> root at ip-172-31-34-24:/etc/opensips# KISep  2 09:50:05 [147490]
> ERROR:proto_wss:wss_conn_init: no TLS server domain found
> Sep  2 09:50:05 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:50:05 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> LL^C
> root at ip-172-31-34-24:/etc/opensips# vi
> /usr/local/etc/opensips/opensips.cfg
> root at ip-172-31-34-24:/etc/opensips# Sep  2 09:50:44 [147490]
> ERROR:proto_wss:wss_conn_init: no TLS server domain found
> Sep  2 09:50:44 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:50:44 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:51:04 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:51:04 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:51:04 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:51:04 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:51:04 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:51:04 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:51:22 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:51:22 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:51:22 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:51:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:51:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:51:40 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:51:40 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:51:40 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:52:00 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:52:00 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:52:00 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:52:19 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:52:19 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:52:19 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:52:19 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:52:19 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:52:19 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:52:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:52:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:52:39 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:52:39 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:52:39 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:52:57 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:52:57 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:52:57 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:53:16 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:53:16 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:53:16 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:53:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:53:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:53:36 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:53:36 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:53:36 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:53:36 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:53:36 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:53:36 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:53:55 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:53:55 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:53:55 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:54:14 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:54:14 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:54:14 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:54:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:54:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:54:34 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:54:34 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:54:34 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:54:53 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:54:53 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:54:53 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:54:53 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:54:53 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:54:53 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:55:12 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:55:12 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:55:12 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: can't send (#7
> iovec buffers) command to a RTP proxy (111:Connection refused)
> Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: timeout
> waiting reply from a RTP proxy
> Sep  2 09:55:27 [147498] ERROR:rtpengine:send_rtpe_command: proxy <udp:
> 127.0.0.0:60000> does not respond, disable it
> Sep  2 09:55:27 [147498] ERROR:rtpengine:rtpe_test: proxy did not respond
> to ping
> Sep  2 09:55:32 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:55:32 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:55:32 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
> Sep  2 09:55:52 [147490] ERROR:proto_wss:wss_conn_init: no TLS server
> domain found
> Sep  2 09:55:52 [147490] ERROR:core:handle_io: failed to do proto 6
> specific init for conn 0x7f2c4123d130
> Sep  2 09:55:52 [147499] ERROR:proto_wss:ws_conn_clean: Failed to retrieve
> the tls_domain pointer in the SSL struct
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Mon Sep  4 06:18:20 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 4 Sep 2023 11:48:20 +0530
Subject: [OpenSIPS-Users] invalid version 12 for table dialog found,
	expected 11
Message-ID: <CAJh_-cDy83b8MXQpq-M-FFsBh7mNF-EXiJDTvdrn0LXaD4fqYA@mail.gmail.com>

ERROR:core:db_check_table_version: invalid version 12 for table dialog
found, expected 11
Sep  4 06:15:07 [195730] ERROR:dialog:init_dlg_db: error during table
version check.
Sep  4 06:15:07 [195730] ERROR:dialog:mod_init: failed to initialize the DB
support
Sep  4 06:15:07 [195730] ERROR:core:init_mod: failed to initialize module
dialog
Sep  4 06:15:07 [195730] ERROR:core:main: error while initializing modules
Sep  4 06:15:07 [195730] INFO:core:cleanup: cleanup
Sep  4 06:15:07 [195730] NOTICE:core:main: Exiting....
Sep  4 06:15:07 [195728] INFO:core:daemonize: pre-daemon process exiting
with -1
-- 
Regards,
B.Prathibha
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From venefax at gmail.com  Mon Sep  4 11:56:11 2023
From: venefax at gmail.com (Saint Michael)
Date: Mon, 4 Sep 2023 07:56:11 -0400
Subject: [OpenSIPS-Users] DNS question
Message-ID: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>

Using opensips 3.1, I am sending calls to a carrier whose address is
sip.xxxxx.com. I add an entry to /etc/hosts/ that redirects sip.xxxxx.com
to a different address, and I test with "resolvectl query sip.xxxxx.com" an
it works as I need it. But opensips keeps sending calls to the DNS address,
not to my new address.
I tried restarting opensips but the issue does not go away.
I tried restarting systemd-resolved, and nothing
I tried opensipsctl fifo cache_flush and it does hang.

So how does Opensips obtains DNS information?
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From mayamatakeshi at gmail.com  Mon Sep  4 12:40:54 2023
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Mon, 4 Sep 2023 21:40:54 +0900
Subject: [OpenSIPS-Users] What happened to auth_db ha1b?
Message-ID: <CABaNFCb8dWSr-qJufVYM_epjUnMzsGAmRbT4eLVH5S-feaJcvA@mail.gmail.com>

Hi,
I'm comparing these docs:

https://www.opensips.org/html/docs/modules/3.1.x/auth_db.html
https://www.opensips.org/html/docs/modules/3.2.x/auth_db.html

The first one mentions ha1b but the second one doesn't mention it.
But the migration doc for 3.1 to 3.2 doesn't mention neither ha1b nor
auth_db:
https://www.opensips.org/Documentation/Migration-3-1-0-to-3-2-0

After update in production to 3.2, I realized a few SIP clients use
username with domain in the Authorization header and so the ha1b should be
used but how do I specify that?
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From medeanwz at gmail.com  Mon Sep  4 22:47:06 2023
From: medeanwz at gmail.com (M S)
Date: Tue, 5 Sep 2023 00:47:06 +0200
Subject: [OpenSIPS-Users] DNS question
In-Reply-To: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>
References: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>
Message-ID: <CAEZrhoPqf2rJT5FMcmCVVjF5f2_zYMoEo_enhTqPguX0OTwvTQ@mail.gmail.com>

Hi,
Opensips asks system's resolver.
Few things to check, make sure your nsswitch.conf file says "files dns"
which is default, also opensips use_naptr option should be yes (default is
yes).
to test, you can also remove dns from nsswitch.conf to make sure your
computer only resolves locally with hosts file. Lastly, you can also do a
tcpdump port 53 to see when/who asks what from who, to debug the issue.

M

On Mon, Sep 4, 2023 at 1:59 PM Saint Michael <venefax at gmail.com> wrote:

> Using opensips 3.1, I am sending calls to a carrier whose address is
> sip.xxxxx.com. I add an entry to /etc/hosts/ that redirects sip.xxxxx.com
> to a different address, and I test with "resolvectl query sip.xxxxx.com" an
> it works as I need it. But opensips keeps sending calls to the DNS address,
> not to my new address.
> I tried restarting opensips but the issue does not go away.
> I tried restarting systemd-resolved, and nothing
> I tried opensipsctl fifo cache_flush and it does hang.
>
> So how does Opensips obtains DNS information?
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From mayamatakeshi at gmail.com  Tue Sep  5 00:25:34 2023
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 5 Sep 2023 09:25:34 +0900
Subject: [OpenSIPS-Users] What happened to auth_db ha1b?
In-Reply-To: <CABaNFCb8dWSr-qJufVYM_epjUnMzsGAmRbT4eLVH5S-feaJcvA@mail.gmail.com>
References: <CABaNFCb8dWSr-qJufVYM_epjUnMzsGAmRbT4eLVH5S-feaJcvA@mail.gmail.com>
Message-ID: <CABaNFCaZuycv+Etkqrb=++90VG=_Tz15q5JAJ+7bQ16+4i+y5g@mail.gmail.com>

On Mon, Sep 4, 2023 at 9:40 PM mayamatakeshi <mayamatakeshi at gmail.com>
wrote:

> Hi,
> I'm comparing these docs:
>
> https://www.opensips.org/html/docs/modules/3.1.x/auth_db.html
> https://www.opensips.org/html/docs/modules/3.2.x/auth_db.html
>
> The first one mentions ha1b but the second one doesn't mention it.
> But the migration doc for 3.1 to 3.2 doesn't mention neither ha1b nor
> auth_db:
> https://www.opensips.org/Documentation/Migration-3-1-0-to-3-2-0
>
> After update in production to 3.2, I realized a few SIP clients use
> username with domain in the Authorization header and so the ha1b should be
> used but how do I specify that?
>
>
Ah. OK. I have found the explanation here:
https://opensips.org/pipermail/users/2021-August/045046.html
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From kwem at gmx.de  Tue Sep  5 07:38:01 2023
From: kwem at gmx.de (Karsten Wemheuer)
Date: Tue, 05 Sep 2023 09:38:01 +0200
Subject: [OpenSIPS-Users] DNS question
In-Reply-To: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>
References: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>
Message-ID: <c38525dc8e2712a5b878e2376efd42352aa99116.camel@gmx.de>

Hi,

Am Montag, dem 04.09.2023 um 07:56 -0400 schrieb Saint Michael:
> Using opensips 3.1, I am sending calls to a carrier whose address is
> sip.xxxxx.com. I add an entry to /etc/hosts/ that redirects
> sip.xxxxx.com to a different address, and I test with "resolvectl
> query sip.xxxxx.com" an it works as I need it. But opensips keeps
> sending calls to the DNS address, not to my new address.
> I tried restarting opensips but the issue does not go away.
> I tried restarting systemd-resolved, and nothing
> I tried opensipsctl fifo cache_flush and it does hang.
>
> So how does Opensips obtains DNS information?

Beside the system resolver also check the settings of the systemd
resolver (/etc/systemd/resolver.conf) (if systemd is in use). There is
a setting to ignore the hosts file, if I remember correctly.

HTH,
Karsten



From venefax at gmail.com  Tue Sep  5 08:21:55 2023
From: venefax at gmail.com (Saint Michael)
Date: Tue, 5 Sep 2023 04:21:55 -0400
Subject: [OpenSIPS-Users] DNS question
In-Reply-To: <c38525dc8e2712a5b878e2376efd42352aa99116.camel@gmx.de>
References: <CAC9cSOB02v_00pbp2-k7jx61h6-QD0Prb1=qnEoKom2N_J-9dQ@mail.gmail.com>
 <c38525dc8e2712a5b878e2376efd42352aa99116.camel@gmx.de>
Message-ID: <CAC9cSOBq9KvPSFU+3SEvQQtnKNta1En3435UW+OXQ2hCRgZF_Q@mail.gmail.com>

The issue is: opensips uses
dig  -t NAPTR sip.xxxxx.com
so you need do that first and get the real terminating hosts and add them
to /etc/hosts
then, it works


On Tue, Sep 5, 2023 at 3:43 AM Karsten Wemheuer <kwem at gmx.de> wrote:

> Hi,
>
> Am Montag, dem 04.09.2023 um 07:56 -0400 schrieb Saint Michael:
> > Using opensips 3.1, I am sending calls to a carrier whose address is
> > sip.xxxxx.com. I add an entry to /etc/hosts/ that redirects
> > sip.xxxxx.com to a different address, and I test with "resolvectl
> > query sip.xxxxx.com" an it works as I need it. But opensips keeps
> > sending calls to the DNS address, not to my new address.
> > I tried restarting opensips but the issue does not go away.
> > I tried restarting systemd-resolved, and nothing
> > I tried opensipsctl fifo cache_flush and it does hang.
> >
> > So how does Opensips obtains DNS information?
>
> Beside the system resolver also check the settings of the systemd
> resolver (/etc/systemd/resolver.conf) (if systemd is in use). There is
> a setting to ignore the hosts file, if I remember correctly.
>
> HTH,
> Karsten
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From tahir at ictinnovations.com  Tue Sep  5 12:22:42 2023
From: tahir at ictinnovations.com (Tahir Almas Dhesi)
Date: Tue, 5 Sep 2023 17:22:42 +0500
Subject: [OpenSIPS-Users] opensips deployment as load balancer for T.38
	traffic
Message-ID: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>

Interested to know how we can deploy opensips as load balancer for  T.38
calls , whether it will work fine or we need to use rabbitmq for load
balancing

scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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From Ben.Newlin at genesys.com  Tue Sep  5 13:34:45 2023
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Tue, 5 Sep 2023 13:34:45 +0000
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS
	library	module loaded
In-Reply-To: <CAJh_-cBoMSqn7w5X0dpWAdd=vaww2UgJ06qg-KKzBvxHs55Y2Q@mail.gmail.com>
References: <CAJh_-cBoMSqn7w5X0dpWAdd=vaww2UgJ06qg-KKzBvxHs55Y2Q@mail.gmail.com>
Message-ID: <BN0PR10MB5208B45552197D2F472E41F095E8A@BN0PR10MB5208.namprd10.prod.outlook.com>

The error is quite clear in the logs you provided:

ERROR:tls_mgm:load_tls_library: No TLS library module loaded

You must load a TLS library in order to use the tls_mgm module.

https://opensips.org/docs/modules/3.4.x/tls_mgm.html#idp5522064
https://opensips.org/docs/modules/3.4.x/tls_openssl.html
https://opensips.org/docs/modules/3.4.x/tls_wolfssl.html

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Prathibha B <prathibhab.tvm at gmail.com>
Date: Friday, September 1, 2023 at 8:12 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS library module loaded
 EXTERNAL EMAIL - Please use caution with links and attachments

________________________________
I've loaded tls_mgm module in /lib64/opensips/modules. Still getting this error.

NOTICE:core:main: version: opensips 3.5.0-dev (x86_64/linux)
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: using 128 MB of shared memory, allocator: Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: using 4 MB of private process memory, allocator: Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:signaling:mod_init: initializing module ...
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:tls_mgm:load_tls_library: No TLS library module loaded
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:core:init_mod: failed to initialize module tls_mgm
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:core:main: failed to initialize modules!
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: Exiting...

--
Regards,
B.Prathibha
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From trevor at webon.co.za  Tue Sep  5 21:49:23 2023
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Tue, 05 Sep 2023 23:49:23 +0200
Subject: [OpenSIPS-Users] Opensips-cp php versions supported
Message-ID: <75cfd12b6fa6fb2210f0aad65e917b68147c639c.camel@webon.co.za>

Hi Guys, 

Cant seem to find any info on what versions of PHP work with OCP, I
just installed 9.3.2 on PHP 8.1 (Centos9 Stream) and other than login
nothing was working every page would just display a blank body,
checking php logs gives me a php exception every page load, switcing to
php7.4 everything seems to be working so far. does OCP not support php
8.X?

[05-Sep-2023 21:27:51 UTC] PHP Fatal error:  Uncaught Error: Call to
undefined function each() in /var/www/opensips-
cp/web/tools/admin/boxes_config/template/menu.php:46


[05-Sep-2023 21:25:42 UTC] PHP Fatal error:  Uncaught Error: Attempt to
assign property "addresses" on null in /var/www/opensips-
cp/config/tools/system/addresses/settings.inc.php:24

Thanks
Trevor


From prathibhab.tvm at gmail.com  Wed Sep  6 04:10:18 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Wed, 6 Sep 2023 04:10:18 +0000
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No
	TLS	library	module loaded
In-Reply-To: <BN0PR10MB5208B45552197D2F472E41F095E8A@BN0PR10MB5208.namprd10.prod.outlook.com>
References: <CAJh_-cBoMSqn7w5X0dpWAdd=vaww2UgJ06qg-KKzBvxHs55Y2Q@mail.gmail.com>
 <BN0PR10MB5208B45552197D2F472E41F095E8A@BN0PR10MB5208.namprd10.prod.outlook.com>
Message-ID: <KL1PR0601MB40042532727275C6665C129AFCEFA@KL1PR0601MB4004.apcprd06.prod.outlook.com>

How to write dialplan in opensips?

Sent from Outlook for Android<https://aka.ms/AAb9ysg>
________________________________
From: Users <users-bounces at lists.opensips.org> on behalf of Ben Newlin <Ben.Newlin at genesys.com>
Sent: Tuesday, September 5, 2023 7:04:45 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS library module loaded


The error is quite clear in the logs you provided:



ERROR:tls_mgm:load_tls_library: No TLS library module loaded



You must load a TLS library in order to use the tls_mgm module.



https://opensips.org/docs/modules/3.4.x/tls_mgm.html#idp5522064

https://opensips.org/docs/modules/3.4.x/tls_openssl.html

https://opensips.org/docs/modules/3.4.x/tls_wolfssl.html



Ben Newlin



From: Users <users-bounces at lists.opensips.org> on behalf of Prathibha B <prathibhab.tvm at gmail.com>
Date: Friday, September 1, 2023 at 8:12 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS library module loaded

 EXTERNAL EMAIL - Please use caution with links and attachments



________________________________

I've loaded tls_mgm module in /lib64/opensips/modules. Still getting this error.

NOTICE:core:main: version: opensips 3.5.0-dev (x86_64/linux)
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: using 128 MB of shared memory, allocator: Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: using 4 MB of private process memory, allocator: Q_MALLOC_DBG
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:signaling:mod_init: initializing module ...
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:tls_mgm:load_tls_library: No TLS library module loaded
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:core:init_mod: failed to initialize module tls_mgm
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] ERROR:core:main: failed to initialize modules!
Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797] NOTICE:core:main: Exiting...



--

Regards,

B.Prathibha
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From rvg at rvgeerligs.nl  Wed Sep  6 05:25:18 2023
From: rvg at rvgeerligs.nl (rvg at rvgeerligs.nl)
Date: Wed, 06 Sep 2023 05:25:18 +0000
Subject: [OpenSIPS-Users] Opensips-cp php versions supported
In-Reply-To: <75cfd12b6fa6fb2210f0aad65e917b68147c639c.camel@webon.co.za>
References: <75cfd12b6fa6fb2210f0aad65e917b68147c639c.camel@webon.co.za>
Message-ID: <d442136647aefb7ac345e7e05a7e0343d03afd4d@rvgeerligs.nl>

Hi,

I had the same. Had to work back towards PHP 7.4. There are very big changes in syntax in 8.1. And it seems the OCP has not adapted its sources. As you have also noticed, that is a lot of work.

Regards,

Ronald



September 5, 2023 at 11:49 PM, trevor at webon.co.za wrote:


> 
> Hi Guys, 
> 
> Cant seem to find any info on what versions of PHP work with OCP, I
> just installed 9.3.2 on PHP 8.1 (Centos9 Stream) and other than login
> nothing was working every page would just display a blank body,
> checking php logs gives me a php exception every page load, switcing to
> php7.4 everything seems to be working so far. does OCP not support php
> 8.X?
> 
> [05-Sep-2023 21:27:51 UTC] PHP Fatal error: Uncaught Error: Call to
> undefined function each() in /var/www/opensips-
> cp/web/tools/admin/boxes_config/template/menu.php:46
> 
> [05-Sep-2023 21:25:42 UTC] PHP Fatal error: Uncaught Error: Attempt to
> assign property "addresses" on null in /var/www/opensips-
> cp/config/tools/system/addresses/settings.inc.php:24
> 
> Thanks
> Trevor
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From trevor at webon.co.za  Wed Sep  6 07:12:48 2023
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Wed, 06 Sep 2023 09:12:48 +0200
Subject: [OpenSIPS-Users] Opensips-cp php versions supported
In-Reply-To: <d442136647aefb7ac345e7e05a7e0343d03afd4d@rvgeerligs.nl>
References: <75cfd12b6fa6fb2210f0aad65e917b68147c639c.camel@webon.co.za>
 <d442136647aefb7ac345e7e05a7e0343d03afd4d@rvgeerligs.nl>
Message-ID: <b590cf04324e13e88f40b0ef8b2c12ee44728517.camel@webon.co.za>

On Wed, 2023-09-06 at 05:25 +0000, rvg at rvgeerligs.nl wrote:
> Hi,
> 
> I had the same. Had to work back towards PHP 7.4. There are very big
> changes in syntax in 8.1. And it seems the OCP has not adapted its
> sources. As you have also noticed, that is a lot of work.
> 
> Regards,
> 
> Ronald

Thanks for the confirmation will have to stick to 7.4 then


From trevor at webon.co.za  Wed Sep  6 11:01:14 2023
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Wed, 06 Sep 2023 13:01:14 +0200
Subject: [OpenSIPS-Users] Yum and Apt repos are down
Message-ID: <5fdb7f821d57c12a001364b7353dc5381060aeb0.camel@webon.co.za>

Hi All, 

repo was working for me last night this morning I am getting 502 Bad
Gateway from both?

https://apt.opensips.org
https://yum.opensips.org


Regards
Trevor 


From mickael at winlux.fr  Wed Sep  6 12:16:45 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 14:16:45 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
Message-ID: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>

Hi all,
I have an issue, when I verify a call with no E164 format (dest:
+3310200123456789)

*logs:*
Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
3310200123456789
Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
(3310200123456789)

*My configuration:*
# ----------------- module  stir_shaken ---------------
loadmodule "stir_shaken.so"
#----------- stir_shaken params -----------------
modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
modparam("stir_shaken", "require_date_hdr", 0)
modparam("stir_shaken", "verify_date_freshness", 60)

According to the doc e164_strict_mode is disabled by default, so I don't
know why it doesn't work.

*source of code: *
        if (_is_e164(num, e164_strict_mode) == -1) {
                LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
num->len, num->s);
                return -1;
        }


Do you have any help for me please ? I have to validate this format of dest
number.

Thanks in advance
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From marcin at voipplus.net  Wed Sep  6 12:21:48 2023
From: marcin at voipplus.net (Marcin Groszek)
Date: Wed, 6 Sep 2023 07:21:48 -0500
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
Message-ID: <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>

Your number is to long

E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting 
with 0

On 9/6/2023 7:16 AM, Mickael Hubert wrote:
> Hi all,
> I have an issue, when I verify a call with no E164 format (dest: 
> +3310200123456789)
>
> _*logs:*_
> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: 
> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 
> format: 3310200123456789
> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: 
> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number 
> (3310200123456789)
>
> _*My configuration:*_
> # ----------------- module  stir_shaken ---------------
> loadmodule "stir_shaken.so"
> #----------- stir_shaken params -----------------
> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
> modparam("stir_shaken", "require_date_hdr", 0)
> modparam("stir_shaken", "verify_date_freshness", 60)
>
> According to the doc e164_strict_mode is disabled by default, so I 
> don't know why it doesn't work.
>
> _*source of code: *_
>         if (_is_e164(num, e164_strict_mode) == -1) {
>                 LM_GEN(log_lev, "number is not in E.164 format: 
> %.*s\n", num->len, num->s);
>                 return -1;
>         }
>
>
> Do you have any help for me please ? I have to validate this format of 
> dest number.
>
> Thanks in advance
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net

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From marcin at voipplus.net  Wed Sep  6 12:23:26 2023
From: marcin at voipplus.net (Marcin Groszek)
Date: Wed, 6 Sep 2023 07:23:26 -0500
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
Message-ID: <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>

Correction : maximum of 15 digits .

On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>
> Your number is to long
>
> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT 
> starting with 0
>
> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>> Hi all,
>> I have an issue, when I verify a call with no E164 format (dest: 
>> +3310200123456789)
>>
>> _*logs:*_
>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: 
>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 
>> format: 3310200123456789
>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: 
>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination 
>> number (3310200123456789)
>>
>> _*My configuration:*_
>> # ----------------- module  stir_shaken ---------------
>> loadmodule "stir_shaken.so"
>> #----------- stir_shaken params -----------------
>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>> modparam("stir_shaken", "require_date_hdr", 0)
>> modparam("stir_shaken", "verify_date_freshness", 60)
>>
>> According to the doc e164_strict_mode is disabled by default, so I 
>> don't know why it doesn't work.
>>
>> _*source of code: *_
>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>                 LM_GEN(log_lev, "number is not in E.164 format: 
>> %.*s\n", num->len, num->s);
>>                 return -1;
>>         }
>>
>>
>> Do you have any help for me please ? I have to validate this format 
>> of dest number.
>>
>> Thanks in advance
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> -- 
> Best Regards:
> Marcin Groszek
> Business Phone Service
> https://www.voipplus.net
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Best Regards:
Marcin Groszek
Business Phone Service
https://www.voipplus.net

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From mickael at winlux.fr  Wed Sep  6 12:26:42 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 14:26:42 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
Message-ID: <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>

yep I found...

if (end - start < 2 || end - start > 15)
return -1;

I have to modify this code.
I will propose a PR.

Thanks a lot
++

Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a écrit :

> Correction : maximum of 15 digits .
> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>
> Your number is to long
>
> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting
> with 0
> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>
> Hi all,
> I have an issue, when I verify a call with no E164 format (dest:
> +3310200123456789)
>
> *logs:*
> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
> 3310200123456789
> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
> (3310200123456789)
>
> *My configuration:*
> # ----------------- module  stir_shaken ---------------
> loadmodule "stir_shaken.so"
> #----------- stir_shaken params -----------------
> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
> modparam("stir_shaken", "require_date_hdr", 0)
> modparam("stir_shaken", "verify_date_freshness", 60)
>
> According to the doc e164_strict_mode is disabled by default, so I don't
> know why it doesn't work.
>
> *source of code: *
>         if (_is_e164(num, e164_strict_mode) == -1) {
>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
> num->len, num->s);
>                 return -1;
>         }
>
>
> Do you have any help for me please ? I have to validate this format of
> dest number.
>
> Thanks in advance
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Best Regards:
> Marcin Groszek
> Business Phone Servicehttps://www.voipplus.net
>
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Best Regards:
> Marcin Groszek
> Business Phone Servicehttps://www.voipplus.net
>
> _______________________________________________
> 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  Wed Sep  6 12:28:51 2023
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Sep 2023 14:28:51 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
Message-ID: <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>

Is ST/SH being used other than the US? AFAIK it only applies to US numbers,
thus 10 digits, no?

On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:

> yep I found...
>
> if (end - start < 2 || end - start > 15)
> return -1;
>
> I have to modify this code.
> I will propose a PR.
>
> Thanks a lot
> ++
>
> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
> écrit :
>
>> Correction : maximum of 15 digits .
>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>
>> Your number is to long
>>
>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting
>> with 0
>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>
>> Hi all,
>> I have an issue, when I verify a call with no E164 format (dest:
>> +3310200123456789)
>>
>> *logs:*
>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>> 3310200123456789
>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>> (3310200123456789)
>>
>> *My configuration:*
>> # ----------------- module  stir_shaken ---------------
>> loadmodule "stir_shaken.so"
>> #----------- stir_shaken params -----------------
>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>> modparam("stir_shaken", "require_date_hdr", 0)
>> modparam("stir_shaken", "verify_date_freshness", 60)
>>
>> According to the doc e164_strict_mode is disabled by default, so I don't
>> know why it doesn't work.
>>
>> *source of code: *
>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
>> num->len, num->s);
>>                 return -1;
>>         }
>>
>>
>> Do you have any help for me please ? I have to validate this format of
>> dest number.
>>
>> Thanks in advance
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>> Best Regards:
>> Marcin Groszek
>> Business Phone Servicehttps://www.voipplus.net
>>
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> --
>> Best Regards:
>> Marcin Groszek
>> Business Phone Servicehttps://www.voipplus.net
>>
>> _______________________________________________
>> 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 mickael at winlux.fr  Wed Sep  6 12:37:50 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 14:37:50 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
Message-ID: <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>

We are deploying it in France.
In France on providers interconnections, we can see a format (made in
France maybe ;) )
prefix: +33
portability prefix: 10200
phonenumber national format without 0: 123456789

++


Le mer. 6 sept. 2023 à 14:30, David Villasmil <
david.villasmil.work at gmail.com> a écrit :

> Is ST/SH being used other than the US? AFAIK it only applies to US
> numbers, thus 10 digits, no?
>
> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>
>> yep I found...
>>
>> if (end - start < 2 || end - start > 15)
>> return -1;
>>
>> I have to modify this code.
>> I will propose a PR.
>>
>> Thanks a lot
>> ++
>>
>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>> écrit :
>>
>>> Correction : maximum of 15 digits .
>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>
>>> Your number is to long
>>>
>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting
>>> with 0
>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>
>>> Hi all,
>>> I have an issue, when I verify a call with no E164 format (dest:
>>> +3310200123456789)
>>>
>>> *logs:*
>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>> 3310200123456789
>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>> (3310200123456789)
>>>
>>> *My configuration:*
>>> # ----------------- module  stir_shaken ---------------
>>> loadmodule "stir_shaken.so"
>>> #----------- stir_shaken params -----------------
>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>> modparam("stir_shaken", "require_date_hdr", 0)
>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>
>>> According to the doc e164_strict_mode is disabled by default, so I
>>> don't know why it doesn't work.
>>>
>>> *source of code: *
>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
>>> num->len, num->s);
>>>                 return -1;
>>>         }
>>>
>>>
>>> Do you have any help for me please ? I have to validate this format of
>>> dest number.
>>>
>>> Thanks in advance
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Best Regards:
>>> Marcin Groszek
>>> Business Phone Servicehttps://www.voipplus.net
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Best Regards:
>>> Marcin Groszek
>>> Business Phone Servicehttps://www.voipplus.net
>>>
>>> _______________________________________________
>>> 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 darencrew at hotmail.com  Wed Sep  6 12:48:49 2023
From: darencrew at hotmail.com (Daren FERREIRA)
Date: Wed, 6 Sep 2023 14:48:49 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
Message-ID: <DM8PR02MB8155BB07B6D5F7B3EEC5CA07B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>

Hello,

Validation of TN should be made without portability prefixes ;)



> Le 6 sept. 2023 à 14:37, Mickael Hubert <mickael at winlux.fr> a écrit :
> 
> We are deploying it in France.
> In France on providers interconnections, we can see a format (made in France maybe ;) )
> prefix: +33
> portability prefix: 10200
> phonenumber national format without 0: 123456789
> 
> ++ <>
> 
> 
> Le mer. 6 sept. 2023 à 14:30, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> a écrit :
>> Is ST/SH being used other than the US? AFAIK it only applies to US numbers, thus 10 digits, no?
>> 
>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr <mailto:mickael at winlux.fr>> wrote:
>>> yep I found...
>>> 
>>> if (end - start < 2 || end - start > 15)
>>> return -1;
>>> 
>>> I have to modify this code.
>>> I will propose a PR.
>>> 
>>> Thanks a lot
>>> ++ <>
>>> 
>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net <mailto:marcin at voipplus.net>> a écrit :
>>>> Correction : maximum of 15 digits .
>>>> 
>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>> Your number is to long
>>>>> 
>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting with 0
>>>>> 
>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>> Hi all,
>>>>>> I have an issue, when I verify a call with no E164 format (dest: +3310200123456789)
>>>>>> 
>>>>>> logs:
>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format: 3310200123456789
>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:w_stir_verify: failed to validate Destination number (3310200123456789)
>>>>>> 
>>>>>> My configuration:
>>>>>> # ----------------- module  stir_shaken ---------------
>>>>>> loadmodule "stir_shaken.so"
>>>>>> #----------- stir_shaken params -----------------
>>>>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>> 
>>>>>> According to the doc e164_strict_mode is disabled by default, so I don't know why it doesn't work.
>>>>>> 
>>>>>> source of code: 
>>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n", num->len, num->s);
>>>>>>                 return -1;
>>>>>>         }
>>>>>> 
>>>>>> 
>>>>>> Do you have any help for me please ? I have to validate this format of dest number.
>>>>>> 
>>>>>> Thanks in advance
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> _______________________________________________
>>>>>> Users mailing list
>>>>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>> -- 
>>>>> Best Regards:
>>>>> Marcin Groszek
>>>>> Business Phone Service
>>>>> https://www.voipplus.net <https://www.voipplus.net/>
>>>>> 
>>>>> _______________________________________________
>>>>> Users mailing list
>>>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>> -- 
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Service
>>>> https://www.voipplus.net <https://www.voipplus.net/>_______________________________________________
>>>> 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 david.villasmil.work at gmail.com  Wed Sep  6 12:50:57 2023
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Wed, 6 Sep 2023 14:50:57 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
Message-ID: <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>

damn... it seems there's a new law in France to do stir/shaken...
Regards,

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


On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr> wrote:

> We are deploying it in France.
> In France on providers interconnections, we can see a format (made in
> France maybe ;) )
> prefix: +33
> portability prefix: 10200
> phonenumber national format without 0: 123456789
>
> ++
>
>
> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
> david.villasmil.work at gmail.com> a écrit :
>
>> Is ST/SH being used other than the US? AFAIK it only applies to US
>> numbers, thus 10 digits, no?
>>
>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>>> yep I found...
>>>
>>> if (end - start < 2 || end - start > 15)
>>> return -1;
>>>
>>> I have to modify this code.
>>> I will propose a PR.
>>>
>>> Thanks a lot
>>> ++
>>>
>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>>> écrit :
>>>
>>>> Correction : maximum of 15 digits .
>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>
>>>> Your number is to long
>>>>
>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>> starting with 0
>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>
>>>> Hi all,
>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>> +3310200123456789)
>>>>
>>>> *logs:*
>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>> 3310200123456789
>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>> (3310200123456789)
>>>>
>>>> *My configuration:*
>>>> # ----------------- module  stir_shaken ---------------
>>>> loadmodule "stir_shaken.so"
>>>> #----------- stir_shaken params -----------------
>>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>
>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>> don't know why it doesn't work.
>>>>
>>>> *source of code: *
>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>> %.*s\n", num->len, num->s);
>>>>                 return -1;
>>>>         }
>>>>
>>>>
>>>> Do you have any help for me please ? I have to validate this format of
>>>> dest number.
>>>>
>>>> Thanks in advance
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>> _______________________________________________
>>>> 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 mickael at winlux.fr  Wed Sep  6 13:09:15 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 15:09:15 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
Message-ID: <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>

Nop Daren, in France it's possible to sign with a portability prefix :(

Le mer. 6 sept. 2023 à 14:53, David Villasmil <
david.villasmil.work at gmail.com> a écrit :

> damn... it seems there's a new law in France to do stir/shaken...
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> phone: +34669448337
>
>
> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr> wrote:
>
>> We are deploying it in France.
>> In France on providers interconnections, we can see a format (made in
>> France maybe ;) )
>> prefix: +33
>> portability prefix: 10200
>> phonenumber national format without 0: 123456789
>>
>> ++
>>
>>
>> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
>> david.villasmil.work at gmail.com> a écrit :
>>
>>> Is ST/SH being used other than the US? AFAIK it only applies to US
>>> numbers, thus 10 digits, no?
>>>
>>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>>>
>>>> yep I found...
>>>>
>>>> if (end - start < 2 || end - start > 15)
>>>> return -1;
>>>>
>>>> I have to modify this code.
>>>> I will propose a PR.
>>>>
>>>> Thanks a lot
>>>> ++
>>>>
>>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>>>> écrit :
>>>>
>>>>> Correction : maximum of 15 digits .
>>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>>
>>>>> Your number is to long
>>>>>
>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>>> starting with 0
>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>
>>>>> Hi all,
>>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>>> +3310200123456789)
>>>>>
>>>>> *logs:*
>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>>> 3310200123456789
>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>>> (3310200123456789)
>>>>>
>>>>> *My configuration:*
>>>>> # ----------------- module  stir_shaken ---------------
>>>>> loadmodule "stir_shaken.so"
>>>>> #----------- stir_shaken params -----------------
>>>>> modparam("stir_shaken", "ca_list",
>>>>> "/usr/local/etc/opensips/man_ca.pem")
>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>
>>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>>> don't know why it doesn't work.
>>>>>
>>>>> *source of code: *
>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>>> %.*s\n", num->len, num->s);
>>>>>                 return -1;
>>>>>         }
>>>>>
>>>>>
>>>>> Do you have any help for me please ? I have to validate this format of
>>>>> dest number.
>>>>>
>>>>> Thanks in advance
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> --
>>>>> Best Regards:
>>>>> Marcin Groszek
>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>
>>>>> --
>>>>> Best Regards:
>>>>> Marcin Groszek
>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>
>>>>> _______________________________________________
>>>>> 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 wadii at evenmedia.fr  Wed Sep  6 13:14:40 2023
From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia)
Date: Wed, 6 Sep 2023 13:14:40 +0000
Subject: [OpenSIPS-Users] =?utf-8?q?RE=C2=A0=3A__Stir_ans_Shaken_-_number_?=
 =?utf-8?q?is_not_in_E=2E164_format?=
In-Reply-To: <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
 <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
Message-ID: <MR1P264MB35859B8D2D5C18EB110E6D61CEEFA@MR1P264MB3585.FRAP264.PROD.OUTLOOK.COM>

Hello Michael,

Are you sure about that ?

As per the French implementation, it's essential to ensure that when an operator processes an Identity Shaken call, they do not modify the "User" section of the SIP From, To, and PAI headers.

Regarding portability, it's important to note that any addition of a portability prefix should exclusively occur at the SIP Request-URI header level.

Source Doc: MAN_Regles-techniques / 2.5.3 Persistance du header Identity.

In simpler terms, this means that the openips stir_shaken_verify function should validate the telephone number found in the TO header which must be in the E.164 format. Any necessary additional prefix should only be present within the RURI

De : Mickael Hubert<mailto:mickael at winlux.fr>
Envoyé le :Wednesday, September 6, 2023 3:12 PM
À : OpenSIPS users mailling list<mailto:users at lists.opensips.org>
Objet :Re: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format

Nop Daren, in France it's possible to sign with a portability prefix :(

Le mer. 6 sept. 2023 à 14:53, David Villasmil <david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>> a écrit :
damn... it seems there's a new law in France to do stir/shaken...
Regards,

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


On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr<mailto:mickael at winlux.fr>> wrote:
We are deploying it in France.
In France on providers interconnections, we can see a format (made in France maybe ;) )
prefix: +33
portability prefix: 10200
phonenumber national format without 0: 123456789

++


Le mer. 6 sept. 2023 à 14:30, David Villasmil <david.villasmil.work at gmail.com<mailto:david.villasmil.work at gmail.com>> a écrit :
Is ST/SH being used other than the US? AFAIK it only applies to US numbers, thus 10 digits, no?

On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr<mailto:mickael at winlux.fr>> wrote:
yep I found...

if (end - start < 2 || end - start > 15)
return -1;

I have to modify this code.
I will propose a PR.

Thanks a lot
++

Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net<mailto:marcin at voipplus.net>> a écrit :

Correction : maximum of 15 digits .
On 9/6/2023 7:21 AM, Marcin Groszek wrote:

Your number is to long

E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting with 0
On 9/6/2023 7:16 AM, Mickael Hubert wrote:
Hi all,
I have an issue, when I verify a call with no E164 format (dest: +3310200123456789)

logs:
Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format: 3310200123456789
Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:w_stir_verify: failed to validate Destination number (3310200123456789)

My configuration:
# ----------------- module  stir_shaken ---------------
loadmodule "stir_shaken.so"
#----------- stir_shaken params -----------------
modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
modparam("stir_shaken", "require_date_hdr", 0)
modparam("stir_shaken", "verify_date_freshness", 60)

According to the doc e164_strict_mode is disabled by default, so I don't know why it doesn't work.

source of code:
        if (_is_e164(num, e164_strict_mode) == -1) {
                LM_GEN(log_lev, "number is not in E.164 format: %.*s\n", num->len, num->s);
                return -1;
        }


Do you have any help for me please ? I have to validate this format of dest number.

Thanks in advance




_______________________________________________

Users mailing list

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

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

--

Best Regards:

Marcin Groszek

Business Phone Service

https://www.voipplus.net



_______________________________________________

Users mailing list

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

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

--

Best Regards:

Marcin Groszek

Business Phone Service

https://www.voipplus.net
_______________________________________________
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<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 darencrew at hotmail.com  Wed Sep  6 13:21:52 2023
From: darencrew at hotmail.com (Daren FERREIRA)
Date: Wed, 6 Sep 2023 15:21:52 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
 <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
Message-ID: <DM8PR02MB81557C97461C69379C91F0A8B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>

We don’t have the same understanding of MAN and FFT rules as, for portability, only R-URI is changed, not the To, that should point to the called number.
And, as To is the source of the TN, the TN shouldn’t contain the portability prefix, and, then, we don’t exceed the 15 numbers limit and E164.

Then, MAN documentation always mention E164 conformity. One of the main rules of E164 is the maximum of 15 digits, so...

So, it’s up to you to make things works as you intend to do. Good luck ;)



Some extracts from MAN documentation as proofs :


Règles techniques:
Remarque sur la portabilité : l’ajout d’un préfixe de portabilité doit être fait uniquement au niveau de l’en-tête SIP Request-URI.


Cahier de tests:

Appel (fixe ou mobile) depuis un ORT1 vers un ORT2 en transit SIP qui retransmet vers ORT3 avec présence de header Identity valide. ORT2 ajoute un préfixe de portabilité pour ORT3 dans R-URI mais pas dans TO (TO n'est pas modifié)



> Le 6 sept. 2023 à 15:09, Mickael Hubert <mickael at winlux.fr> a écrit :
> 
> Nop Daren, in France it's possible to sign with a portability prefix :(
> 
> Le mer. 6 sept. 2023 à 14:53, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> a écrit :
>> damn... it seems there's a new law in France to do stir/shaken... 
>> Regards,
>> 
>> David Villasmil
>> email: david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>
>> phone: +34669448337
>> 
>> 
>> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr <mailto:mickael at winlux.fr>> wrote:
>>> We are deploying it in France.
>>> In France on providers interconnections, we can see a format (made in France maybe ;) )
>>> prefix: +33
>>> portability prefix: 10200
>>> phonenumber national format without 0: 123456789
>>> 
>>> ++ <>
>>> 
>>> 
>>> Le mer. 6 sept. 2023 à 14:30, David Villasmil <david.villasmil.work at gmail.com <mailto:david.villasmil.work at gmail.com>> a écrit :
>>>> Is ST/SH being used other than the US? AFAIK it only applies to US numbers, thus 10 digits, no?
>>>> 
>>>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr <mailto:mickael at winlux.fr>> wrote:
>>>>> yep I found...
>>>>> 
>>>>> if (end - start < 2 || end - start > 15)
>>>>> return -1;
>>>>> 
>>>>> I have to modify this code.
>>>>> I will propose a PR.
>>>>> 
>>>>> Thanks a lot
>>>>> ++ <>
>>>>> 
>>>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net <mailto:marcin at voipplus.net>> a écrit :
>>>>>> Correction : maximum of 15 digits .
>>>>>> 
>>>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>>>> Your number is to long
>>>>>>> 
>>>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting with 0
>>>>>>> 
>>>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>>>> Hi all,
>>>>>>>> I have an issue, when I verify a call with no E164 format (dest: +3310200123456789)
>>>>>>>> 
>>>>>>>> logs:
>>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format: 3310200123456789
>>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]: ERROR:stir_shaken:w_stir_verify: failed to validate Destination number (3310200123456789)
>>>>>>>> 
>>>>>>>> My configuration:
>>>>>>>> # ----------------- module  stir_shaken ---------------
>>>>>>>> loadmodule "stir_shaken.so"
>>>>>>>> #----------- stir_shaken params -----------------
>>>>>>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>>>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>>>> 
>>>>>>>> According to the doc e164_strict_mode is disabled by default, so I don't know why it doesn't work.
>>>>>>>> 
>>>>>>>> source of code: 
>>>>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>>>>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n", num->len, num->s);
>>>>>>>>                 return -1;
>>>>>>>>         }
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Do you have any help for me please ? I have to validate this format of dest number.
>>>>>>>> 
>>>>>>>> Thanks in advance
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing list
>>>>>>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>> -- 
>>>>>>> Best Regards:
>>>>>>> Marcin Groszek
>>>>>>> Business Phone Service
>>>>>>> https://www.voipplus.net <https://www.voipplus.net/>
>>>>>>> 
>>>>>>> _______________________________________________
>>>>>>> Users mailing list
>>>>>>> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>>>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>> -- 
>>>>>> Best Regards:
>>>>>> Marcin Groszek
>>>>>> Business Phone Service
>>>>>> https://www.voipplus.net <https://www.voipplus.net/>_______________________________________________
>>>>>> 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 <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 mickael at winlux.fr  Wed Sep  6 13:35:06 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 15:35:06 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <DM8PR02MB81557C97461C69379C91F0A8B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
 <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
 <DM8PR02MB81557C97461C69379C91F0A8B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>
Message-ID: <CA+fRVSXK7vmqAiyCWJAa1qvOn6xgCa4G2cXjegDi6y1iWt37nA@mail.gmail.com>

Thanks a lot Daren,
I have to contact this big french provider to explain its issue ;)

Le mer. 6 sept. 2023 à 15:24, Daren FERREIRA <darencrew at hotmail.com> a
écrit :

> We don’t have the same understanding of MAN and FFT rules as, for
> portability, only R-URI is changed, not the To, that should point to the
> called number.
> And, as To is the source of the TN, the TN shouldn’t contain the
> portability prefix, and, then, we don’t exceed the 15 numbers limit and
> E164.
>
> Then, MAN documentation always mention E164 conformity. One of the main
> rules of E164 is the maximum of 15 digits, so...
>
> So, it’s up to you to make things works as you intend to do. Good luck ;)
>
>
>
> Some extracts from MAN documentation as proofs :
>
>
> Règles techniques:
>
> Remarque sur la portabilité : l’ajout d’un préfixe de portabilité doit
> être fait uniquement au niveau de l’en-tête SIP Request-URI.
>
> Cahier de tests:
>
> Appel (fixe ou mobile) depuis un ORT1 vers un ORT2 en transit SIP qui
> retransmet vers ORT3 avec présence de header Identity valide. ORT2 ajoute
> un préfixe de portabilité pour ORT3 dans R-URI mais pas dans TO (TO n'est
> pas modifié)
>
>
>
> Le 6 sept. 2023 à 15:09, Mickael Hubert <mickael at winlux.fr> a écrit :
>
> Nop Daren, in France it's possible to sign with a portability prefix :(
>
> Le mer. 6 sept. 2023 à 14:53, David Villasmil <
> david.villasmil.work at gmail.com> a écrit :
>
>> damn... it seems there's a new law in France to do stir/shaken...
>> Regards,
>>
>> David Villasmil
>> email: david.villasmil.work at gmail.com
>> phone: +34669448337
>>
>>
>> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr> wrote:
>>
>>> We are deploying it in France.
>>> In France on providers interconnections, we can see a format (made in
>>> France maybe ;) )
>>> prefix: +33
>>> portability prefix: 10200
>>> phonenumber national format without 0: 123456789
>>>
>>> ++
>>>
>>>
>>> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
>>> david.villasmil.work at gmail.com> a écrit :
>>>
>>>> Is ST/SH being used other than the US? AFAIK it only applies to US
>>>> numbers, thus 10 digits, no?
>>>>
>>>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>
>>>>> yep I found...
>>>>>
>>>>> if (end - start < 2 || end - start > 15)
>>>>> return -1;
>>>>>
>>>>> I have to modify this code.
>>>>> I will propose a PR.
>>>>>
>>>>> Thanks a lot
>>>>> ++
>>>>>
>>>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>>>>> écrit :
>>>>>
>>>>>> Correction : maximum of 15 digits .
>>>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>>>
>>>>>> Your number is to long
>>>>>>
>>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>>>> starting with 0
>>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>>
>>>>>> Hi all,
>>>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>>>> +3310200123456789)
>>>>>>
>>>>>> *logs:*
>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>>>> 3310200123456789
>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>>>> (3310200123456789)
>>>>>>
>>>>>> *My configuration:*
>>>>>> # ----------------- module  stir_shaken ---------------
>>>>>> loadmodule "stir_shaken.so"
>>>>>> #----------- stir_shaken params -----------------
>>>>>> modparam("stir_shaken", "ca_list",
>>>>>> "/usr/local/etc/opensips/man_ca.pem")
>>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>>
>>>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>>>> don't know why it doesn't work.
>>>>>>
>>>>>> *source of code: *
>>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>>>> %.*s\n", num->len, num->s);
>>>>>>                 return -1;
>>>>>>         }
>>>>>>
>>>>>>
>>>>>> Do you have any help for me please ? I have to validate this format
>>>>>> of dest number.
>>>>>>
>>>>>> Thanks in advance
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>> --
>>>>>> Best Regards:
>>>>>> Marcin Groszek
>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>
>>>>>> --
>>>>>> Best Regards:
>>>>>> Marcin Groszek
>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>
>>>>>> _______________________________________________
>>>>>> 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
>>
> _______________________________________________
> 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 mickael at winlux.fr  Wed Sep  6 14:07:30 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 16:07:30 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSXK7vmqAiyCWJAa1qvOn6xgCa4G2cXjegDi6y1iWt37nA@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
 <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
 <DM8PR02MB81557C97461C69379C91F0A8B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>
 <CA+fRVSXK7vmqAiyCWJAa1qvOn6xgCa4G2cXjegDi6y1iWt37nA@mail.gmail.com>
Message-ID: <CA+fRVSWU-6CzUajxqhrcA1SzOBFhtOR++FOPzcRdB06Ctd82mg@mail.gmail.com>

Answer of this french provider (in french sorry)

*Pour le 1er point, la clause 2.5.3 des règles techniques MAN est pour le
cas où l’opérateur de transit ajoute un préfixe de portabilité, afin de
préservé l’Identity initial, il est impérative de ne pas modifier le header
To.*

*Par contre, dans le cas où c’est l’opérateur origine d’initie un appel
avec préfixe de portabilité, il n’est pas interdit de valoriser le header
To avec le préfixe de portabilité, à condition que le Token contient le
claim « dest » égale la valeur du header To. Le résultat de vérification
devra être OK.*


What do you think ?

Le mer. 6 sept. 2023 à 15:35, Mickael Hubert <mickael at winlux.fr> a écrit :

> Thanks a lot Daren,
> I have to contact this big french provider to explain its issue ;)
>
> Le mer. 6 sept. 2023 à 15:24, Daren FERREIRA <darencrew at hotmail.com> a
> écrit :
>
>> We don’t have the same understanding of MAN and FFT rules as, for
>> portability, only R-URI is changed, not the To, that should point to the
>> called number.
>> And, as To is the source of the TN, the TN shouldn’t contain the
>> portability prefix, and, then, we don’t exceed the 15 numbers limit and
>> E164.
>>
>> Then, MAN documentation always mention E164 conformity. One of the main
>> rules of E164 is the maximum of 15 digits, so...
>>
>> So, it’s up to you to make things works as you intend to do. Good luck ;)
>>
>>
>>
>> Some extracts from MAN documentation as proofs :
>>
>>
>> Règles techniques:
>>
>> Remarque sur la portabilité : l’ajout d’un préfixe de portabilité
>> doit être fait uniquement au niveau de l’en-tête SIP Request-URI.
>>
>> Cahier de tests:
>>
>> Appel (fixe ou mobile) depuis un ORT1 vers un ORT2 en transit SIP qui
>> retransmet vers ORT3 avec présence de header Identity valide. ORT2
>> ajoute un préfixe de portabilité pour ORT3 dans R-URI mais pas dans TO (TO
>> n'est pas modifié)
>>
>>
>>
>> Le 6 sept. 2023 à 15:09, Mickael Hubert <mickael at winlux.fr> a écrit :
>>
>> Nop Daren, in France it's possible to sign with a portability prefix :(
>>
>> Le mer. 6 sept. 2023 à 14:53, David Villasmil <
>> david.villasmil.work at gmail.com> a écrit :
>>
>>> damn... it seems there's a new law in France to do stir/shaken...
>>> Regards,
>>>
>>> David Villasmil
>>> email: david.villasmil.work at gmail.com
>>> phone: +34669448337
>>>
>>>
>>> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr> wrote:
>>>
>>>> We are deploying it in France.
>>>> In France on providers interconnections, we can see a format (made in
>>>> France maybe ;) )
>>>> prefix: +33
>>>> portability prefix: 10200
>>>> phonenumber national format without 0: 123456789
>>>>
>>>> ++
>>>>
>>>>
>>>> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
>>>> david.villasmil.work at gmail.com> a écrit :
>>>>
>>>>> Is ST/SH being used other than the US? AFAIK it only applies to US
>>>>> numbers, thus 10 digits, no?
>>>>>
>>>>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>>>>>
>>>>>> yep I found...
>>>>>>
>>>>>> if (end - start < 2 || end - start > 15)
>>>>>> return -1;
>>>>>>
>>>>>> I have to modify this code.
>>>>>> I will propose a PR.
>>>>>>
>>>>>> Thanks a lot
>>>>>> ++
>>>>>>
>>>>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>>>>>> écrit :
>>>>>>
>>>>>>> Correction : maximum of 15 digits .
>>>>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>>>>
>>>>>>> Your number is to long
>>>>>>>
>>>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>>>>> starting with 0
>>>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>>>
>>>>>>> Hi all,
>>>>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>>>>> +3310200123456789)
>>>>>>>
>>>>>>> *logs:*
>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>>>>> 3310200123456789
>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>>>>> (3310200123456789)
>>>>>>>
>>>>>>> *My configuration:*
>>>>>>> # ----------------- module  stir_shaken ---------------
>>>>>>> loadmodule "stir_shaken.so"
>>>>>>> #----------- stir_shaken params -----------------
>>>>>>> modparam("stir_shaken", "ca_list",
>>>>>>> "/usr/local/etc/opensips/man_ca.pem")
>>>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>>>
>>>>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>>>>> don't know why it doesn't work.
>>>>>>>
>>>>>>> *source of code: *
>>>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>>>>> %.*s\n", num->len, num->s);
>>>>>>>                 return -1;
>>>>>>>         }
>>>>>>>
>>>>>>>
>>>>>>> Do you have any help for me please ? I have to validate this format
>>>>>>> of dest number.
>>>>>>>
>>>>>>> Thanks in advance
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>> --
>>>>>>> Best Regards:
>>>>>>> Marcin Groszek
>>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>
>>>>>>> --
>>>>>>> Best Regards:
>>>>>>> Marcin Groszek
>>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>
>> _______________________________________________
>> 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 mickael at winlux.fr  Wed Sep  6 14:52:53 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 6 Sep 2023 16:52:53 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CA+fRVSWU-6CzUajxqhrcA1SzOBFhtOR++FOPzcRdB06Ctd82mg@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CA+fRVSVyG+6M6LwjRDnJizpPOFvjA-w47kfZn-M_2JZ3bytPFQ@mail.gmail.com>
 <CAFGRPVoRwkTstZa1y_urW=HNNSSO3aEhX+g==C3qXO_JXkzdzw@mail.gmail.com>
 <CA+fRVSUMWo9_shVBGu2-fHrvo2KjKRvP6BUfT_kQ1G5ABEx=9Q@mail.gmail.com>
 <DM8PR02MB81557C97461C69379C91F0A8B0EFA@DM8PR02MB8155.namprd02.prod.outlook.com>
 <CA+fRVSXK7vmqAiyCWJAa1qvOn6xgCa4G2cXjegDi6y1iWt37nA@mail.gmail.com>
 <CA+fRVSWU-6CzUajxqhrcA1SzOBFhtOR++FOPzcRdB06Ctd82mg@mail.gmail.com>
Message-ID: <CA+fRVSViaM3ppLmcR+=t7vAhkmLdL0WO3YmEDPL_C5TJ+EwuGw@mail.gmail.com>

Final answer (in french):

*Par ailleurs, il semble que ce point n’est pas clair pour tout le monde,
APNF a soulevé ce point ce matin en atelier MAN. Il y aura une mise à jour
du document « Règles techniques MAN » pour bien clarifier afin d’éviter des
mauvaises interprétations. Je t’enverrai le texte clarifié discuté ce matin
dès que je le reçois de la part de APNF.*

So we have to accept more than 15 digits in dest...

Le mer. 6 sept. 2023 à 16:07, Mickael Hubert <mickael at winlux.fr> a écrit :

> Answer of this french provider (in french sorry)
>
> *Pour le 1er point, la clause 2.5.3 des règles techniques MAN est pour le
> cas où l’opérateur de transit ajoute un préfixe de portabilité, afin de
> préservé l’Identity initial, il est impérative de ne pas modifier le header
> To.*
>
> *Par contre, dans le cas où c’est l’opérateur origine d’initie un appel
> avec préfixe de portabilité, il n’est pas interdit de valoriser le header
> To avec le préfixe de portabilité, à condition que le Token contient le
> claim « dest » égale la valeur du header To. Le résultat de vérification
> devra être OK.*
>
>
> What do you think ?
>
> Le mer. 6 sept. 2023 à 15:35, Mickael Hubert <mickael at winlux.fr> a écrit :
>
>> Thanks a lot Daren,
>> I have to contact this big french provider to explain its issue ;)
>>
>> Le mer. 6 sept. 2023 à 15:24, Daren FERREIRA <darencrew at hotmail.com> a
>> écrit :
>>
>>> We don’t have the same understanding of MAN and FFT rules as, for
>>> portability, only R-URI is changed, not the To, that should point to the
>>> called number.
>>> And, as To is the source of the TN, the TN shouldn’t contain the
>>> portability prefix, and, then, we don’t exceed the 15 numbers limit and
>>> E164.
>>>
>>> Then, MAN documentation always mention E164 conformity. One of the main
>>> rules of E164 is the maximum of 15 digits, so...
>>>
>>> So, it’s up to you to make things works as you intend to do. Good luck ;)
>>>
>>>
>>>
>>> Some extracts from MAN documentation as proofs :
>>>
>>>
>>> Règles techniques:
>>>
>>> Remarque sur la portabilité : l’ajout d’un préfixe de portabilité
>>> doit être fait uniquement au niveau de l’en-tête SIP Request-URI.
>>>
>>> Cahier de tests:
>>>
>>> Appel (fixe ou mobile) depuis un ORT1 vers un ORT2 en transit SIP qui
>>> retransmet vers ORT3 avec présence de header Identity valide. ORT2
>>> ajoute un préfixe de portabilité pour ORT3 dans R-URI mais pas dans TO (TO
>>> n'est pas modifié)
>>>
>>>
>>>
>>> Le 6 sept. 2023 à 15:09, Mickael Hubert <mickael at winlux.fr> a écrit :
>>>
>>> Nop Daren, in France it's possible to sign with a portability prefix :(
>>>
>>> Le mer. 6 sept. 2023 à 14:53, David Villasmil <
>>> david.villasmil.work at gmail.com> a écrit :
>>>
>>>> damn... it seems there's a new law in France to do stir/shaken...
>>>> Regards,
>>>>
>>>> David Villasmil
>>>> email: david.villasmil.work at gmail.com
>>>> phone: +34669448337
>>>>
>>>>
>>>> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert <mickael at winlux.fr>
>>>> wrote:
>>>>
>>>>> We are deploying it in France.
>>>>> In France on providers interconnections, we can see a format (made in
>>>>> France maybe ;) )
>>>>> prefix: +33
>>>>> portability prefix: 10200
>>>>> phonenumber national format without 0: 123456789
>>>>>
>>>>> ++
>>>>>
>>>>>
>>>>> Le mer. 6 sept. 2023 à 14:30, David Villasmil <
>>>>> david.villasmil.work at gmail.com> a écrit :
>>>>>
>>>>>> Is ST/SH being used other than the US? AFAIK it only applies to US
>>>>>> numbers, thus 10 digits, no?
>>>>>>
>>>>>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr>
>>>>>> wrote:
>>>>>>
>>>>>>> yep I found...
>>>>>>>
>>>>>>> if (end - start < 2 || end - start > 15)
>>>>>>> return -1;
>>>>>>>
>>>>>>> I have to modify this code.
>>>>>>> I will propose a PR.
>>>>>>>
>>>>>>> Thanks a lot
>>>>>>> ++
>>>>>>>
>>>>>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net>
>>>>>>> a écrit :
>>>>>>>
>>>>>>>> Correction : maximum of 15 digits .
>>>>>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>>>>>
>>>>>>>> Your number is to long
>>>>>>>>
>>>>>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>>>>>> starting with 0
>>>>>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>>>>>
>>>>>>>> Hi all,
>>>>>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>>>>>> +3310200123456789)
>>>>>>>>
>>>>>>>> *logs:*
>>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>>>>>> 3310200123456789
>>>>>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>>>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>>>>>> (3310200123456789)
>>>>>>>>
>>>>>>>> *My configuration:*
>>>>>>>> # ----------------- module  stir_shaken ---------------
>>>>>>>> loadmodule "stir_shaken.so"
>>>>>>>> #----------- stir_shaken params -----------------
>>>>>>>> modparam("stir_shaken", "ca_list",
>>>>>>>> "/usr/local/etc/opensips/man_ca.pem")
>>>>>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>>>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>>>>>
>>>>>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>>>>>> don't know why it doesn't work.
>>>>>>>>
>>>>>>>> *source of code: *
>>>>>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>>>>>> %.*s\n", num->len, num->s);
>>>>>>>>                 return -1;
>>>>>>>>         }
>>>>>>>>
>>>>>>>>
>>>>>>>> Do you have any help for me please ? I have to validate this format
>>>>>>>> of dest number.
>>>>>>>>
>>>>>>>> Thanks in advance
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>> --
>>>>>>>> Best Regards:
>>>>>>>> Marcin Groszek
>>>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>>>
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>>>>>
>>>>>>>> --
>>>>>>>> Best Regards:
>>>>>>>> Marcin Groszek
>>>>>>>> Business Phone Servicehttps://www.voipplus.net
>>>>>>>>
>>>>>>>> _______________________________________________
>>>>>>>> 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
>>>>
>>> _______________________________________________
>>> 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 brett at nemeroff.com  Wed Sep  6 17:21:17 2023
From: brett at nemeroff.com (Brett Nemeroff)
Date: Wed, 6 Sep 2023 12:21:17 -0500
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
Message-ID: <CAPwC5wy1h=ya3q48m5ucNJtfyWdu7nJGVeQrJ9ts=YHeZnedkw@mail.gmail.com>

Buckle up! Things are going to get interesting. This is just one tiny
little example...

We have a lot of international issues when it comes to building trust and
call authentication internationally. For example, consider that STIR/SHAKEN
as we know it is built upon the US policy and governance authority and as
such certificates come from US STI-CA. What will happen internationally
with certificates and trust? Will PASSporTs all look the same? How do we
trust calls that cross borders?

PASSporT formatting is the tip of the iceberg. However for France calling,
they have their own set of standards governed by MAN. OpenSIPs modules
should certainly have a way to support them, but I would be surprised if
they do out of the box. Given how new all of this is, I'd expect we'd need
the help of someone like Mickael to help understand where the module is
lacking for France standards.

For those who are interested, i3Forum is having a talk tomorrow morning
(US) on international calling and trust which may cover some of these
issues. I am not affiliated with that group, but I'll be joining the
webinar to gather information. I'm passing the information on to you guys.

https://i3forum.org/one-consortium/

-Brett


On Wed, Sep 6, 2023 at 7:31 AM David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Is ST/SH being used other than the US? AFAIK it only applies to US
> numbers, thus 10 digits, no?
>
> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>
>> yep I found...
>>
>> if (end - start < 2 || end - start > 15)
>> return -1;
>>
>> I have to modify this code.
>> I will propose a PR.
>>
>> Thanks a lot
>> ++
>>
>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>> écrit :
>>
>>> Correction : maximum of 15 digits .
>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>
>>> Your number is to long
>>>
>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT starting
>>> with 0
>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>
>>> Hi all,
>>> I have an issue, when I verify a call with no E164 format (dest:
>>> +3310200123456789)
>>>
>>> *logs:*
>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>> 3310200123456789
>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>> (3310200123456789)
>>>
>>> *My configuration:*
>>> # ----------------- module  stir_shaken ---------------
>>> loadmodule "stir_shaken.so"
>>> #----------- stir_shaken params -----------------
>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>> modparam("stir_shaken", "require_date_hdr", 0)
>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>
>>> According to the doc e164_strict_mode is disabled by default, so I
>>> don't know why it doesn't work.
>>>
>>> *source of code: *
>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>                 LM_GEN(log_lev, "number is not in E.164 format: %.*s\n",
>>> num->len, num->s);
>>>                 return -1;
>>>         }
>>>
>>>
>>> Do you have any help for me please ? I have to validate this format of
>>> dest number.
>>>
>>> Thanks in advance
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Best Regards:
>>> Marcin Groszek
>>> Business Phone Servicehttps://www.voipplus.net
>>>
>>>
>>> _______________________________________________
>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>> --
>>> Best Regards:
>>> Marcin Groszek
>>> Business Phone Servicehttps://www.voipplus.net
>>>
>>> _______________________________________________
>>> 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 brett at nemeroff.com  Wed Sep  6 17:37:41 2023
From: brett at nemeroff.com (Brett Nemeroff)
Date: Wed, 6 Sep 2023 12:37:41 -0500
Subject: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS library
 module loaded
In-Reply-To: <KL1PR0601MB40042532727275C6665C129AFCEFA@KL1PR0601MB4004.apcprd06.prod.outlook.com>
References: <CAJh_-cBoMSqn7w5X0dpWAdd=vaww2UgJ06qg-KKzBvxHs55Y2Q@mail.gmail.com>
 <BN0PR10MB5208B45552197D2F472E41F095E8A@BN0PR10MB5208.namprd10.prod.outlook.com>
 <KL1PR0601MB40042532727275C6665C129AFCEFA@KL1PR0601MB4004.apcprd06.prod.outlook.com>
Message-ID: <CAPwC5wzJSwUsHsX0ROXhD2cMh-tPFfiFFni4XRqBupSq6B7VLQ@mail.gmail.com>

Hello,
Given your recent flurry of emails to the mailing list, I'm sure you are
learning quickly how to use OpenSIPs. However, I would like to remind you
to please remember proper netiquette when collaborating on the mailing
list. The OpenSIPs community is full of very friendly and helpful members.
Most of us have day jobs and don't babysit the list. That being said, as is
with most technical mailing lists, it is best to clearly and concisely ask
your questions to get the most help and to not waste the time of
contributors on the list. Additionally, it's always good to show that you
"gave it a try yourself" and what did and didn't work when you did. I've
seen a number of your posts that only include error logs or like the above
open ended question that is actually attached to a thread of another
concern.

We are all happy to help, but please start by:
1. Read the documentation
2. Try it out yourself
3. Show what you tried, and what didn't work
4. Include the version number you are doing
5. Give us a little context
6. We are human. Say hello. Be human.
7. Remember that your question may have been asked before and/or others may
have the same question. It's important to keep your questions in threads so
others can also benefit from the list interaction.

Additionally, there is a slack group. It's not super active, but I know
many of us hang out in there (myself included). If you show up, be a
friendly member of the community, and ask a thoughtful question, you are
bound to get more help.

Thanks,
Brett



On Tue, Sep 5, 2023 at 11:12 PM Prathibha B <prathibhab.tvm at gmail.com>
wrote:

> How to write dialplan in opensips?
>
> Sent from Outlook for Android <https://aka.ms/AAb9ysg>
> ------------------------------
> *From:* Users <users-bounces at lists.opensips.org> on behalf of Ben Newlin <
> Ben.Newlin at genesys.com>
> *Sent:* Tuesday, September 5, 2023 7:04:45 PM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject:* Re: [OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS
> library module loaded
>
>
> The error is quite clear in the logs you provided:
>
>
>
> ERROR:tls_mgm:load_tls_library: No TLS library module loaded
>
>
>
> You must load a TLS library in order to use the tls_mgm module.
>
>
>
> https://opensips.org/docs/modules/3.4.x/tls_mgm.html#idp5522064
>
> https://opensips.org/docs/modules/3.4.x/tls_openssl.html
>
> https://opensips.org/docs/modules/3.4.x/tls_wolfssl.html
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Prathibha B
> <prathibhab.tvm at gmail.com>
> *Date: *Friday, September 1, 2023 at 8:12 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] ERROR:tls_mgm:load_tls_library: No TLS
> library module loaded
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> ------------------------------
>
> I've loaded tls_mgm module in /lib64/opensips/modules. Still getting this
> error.
>
> NOTICE:core:main: version: opensips 3.5.0-dev (x86_64/linux)
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> NOTICE:core:main: using 128 MB of shared memory, allocator: Q_MALLOC_DBG
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> NOTICE:core:main: using 4 MB of private process memory, allocator:
> Q_MALLOC_DBG
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> NOTICE:signaling:mod_init: initializing module ...
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> ERROR:tls_mgm:load_tls_library: No TLS library module loaded
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> ERROR:core:init_mod: failed to initialize module tls_mgm
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> ERROR:core:main: failed to initialize modules!
> Sep 01 10:45:49 ip-172-31-34-24 opensips[363797]: Sep 1 10:45:49 [363797]
> NOTICE:core:main: Exiting...
>
>
>
> --
>
> Regards,
>
> B.Prathibha
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From razvan at opensips.org  Thu Sep  7 07:12:18 2023
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Thu, 7 Sep 2023 10:12:18 +0300
Subject: [OpenSIPS-Users] Yum and Apt repos are down
In-Reply-To: <5fdb7f821d57c12a001364b7353dc5381060aeb0.camel@webon.co.za>
References: <5fdb7f821d57c12a001364b7353dc5381060aeb0.camel@webon.co.za>
Message-ID: <6de8fa68-e63f-4294-ad5c-a1ac987e37ad@opensips.org>

This has been fixed, can you please confirm, or let us know if the 
problem still persists for you.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/6/23 14:01, trevor at webon.co.za wrote:
> Hi All,
> 
> repo was working for me last night this morning I am getting 502 Bad
> Gateway from both?
> 
> https://apt.opensips.org
> https://yum.opensips.org
> 
> 
> Regards
> Trevor
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From trevor at webon.co.za  Thu Sep  7 07:18:51 2023
From: trevor at webon.co.za (trevor at webon.co.za)
Date: Thu, 07 Sep 2023 09:18:51 +0200
Subject: [OpenSIPS-Users] Yum and Apt repos are down
In-Reply-To: <6de8fa68-e63f-4294-ad5c-a1ac987e37ad@opensips.org>
References: <5fdb7f821d57c12a001364b7353dc5381060aeb0.camel@webon.co.za>
 <6de8fa68-e63f-4294-ad5c-a1ac987e37ad@opensips.org>
Message-ID: <c5de61627c2911f8366a4ba7920c6a13c7331640.camel@webon.co.za>

On Thu, 2023-09-07 at 10:12 +0300, Răzvan Crainea wrote:
> This has been fixed, can you please confirm, or let us know if the 
> problem still persists for you.
> 
> Best regards,
> 
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
> 
> On 9/6/23 14:01, trevor at webon.co.za wrote:
> 

Hi confirmed its back up thanks so much

Regards, 
Trevor


From johan at democon.be  Thu Sep  7 07:42:46 2023
From: johan at democon.be (johan)
Date: Thu, 7 Sep 2023 09:42:46 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <CAPwC5wy1h=ya3q48m5ucNJtfyWdu7nJGVeQrJ9ts=YHeZnedkw@mail.gmail.com>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CAPwC5wy1h=ya3q48m5ucNJtfyWdu7nJGVeQrJ9ts=YHeZnedkw@mail.gmail.com>
Message-ID: <a4fe9736-462b-03f0-0ca3-79a2eaf2fcbc@democon.be>

I think that having this on a global scale will result in a mumbo jumbo 
of standards and implementations.  Best thing for Europe would be that 
this would be tackled on EU scale, but it don't see this happening (e.g. 
what France does, is for sure not okay for Germany and so on).   Anyway, 
I will be on that call too, as I would like to hear their view on this 
complicated item.

On 6/09/2023 19:21, Brett Nemeroff wrote:
> Buckle up! Things are going to get interesting. This is just one tiny 
> little example...
>
> We have a lot of international issues when it comes to building trust 
> and call authentication internationally. For example, consider that 
> STIR/SHAKEN as we know it is built upon the US policy and governance 
> authority and as such certificates come from US STI-CA. What will 
> happen internationally with certificates and trust? Will PASSporTs all 
> look the same? How do we trust calls that cross borders?
>
> PASSporT formatting is the tip of the iceberg. However for France 
> calling, they have their own set of standards governed by MAN. 
> OpenSIPs modules should certainly have a way to support them, but I 
> would be surprised if they do out of the box. Given how new all of 
> this is, I'd expect we'd need the help of someone like Mickael to help 
> understand where the module is lacking for France standards.
>
> For those who are interested, i3Forum is having a talk tomorrow 
> morning (US) on international calling and trust which may cover some 
> of these issues. I am not affiliated with that group, but I'll be 
> joining the webinar to gather information. I'm passing the information 
> on to you guys.
>
> https://i3forum.org/one-consortium/
>
> -Brett
>
>
> On Wed, Sep 6, 2023 at 7:31 AM David Villasmil 
> <david.villasmil.work at gmail.com> wrote:
>
>     Is ST/SH being used other than the US? AFAIK it only applies to US
>     numbers, thus 10 digits, no?
>
>     On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>
>         yep I found...
>
>         if (end - start < 2 || end - start > 15)
>         return -1;
>
>         I have to modify this code.
>         I will propose a PR.
>
>         Thanks a lot
>         ++
>
>         Le mer. 6 sept. 2023 à 14:25, Marcin Groszek
>         <marcin at voipplus.net> a écrit :
>
>             Correction : maximum of 15 digits .
>
>             On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>
>>             Your number is to long
>>
>>             E.164 is + [1-9]  and  {1-14} digits for total of 15
>>             digits NOT starting with 0
>>
>>             On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>             Hi all,
>>>             I have an issue, when I verify a call with no E164
>>>             format (dest: +3310200123456789)
>>>
>>>             _*logs:*_
>>>             Sep  6 13:39:48 am-scr-001
>>>             /usr/local/sbin/opensips[622409]:
>>>             ERROR:stir_shaken:check_passport_phonenum: number is not
>>>             in E.164 format: 3310200123456789
>>>             Sep  6 13:39:48 am-scr-001
>>>             /usr/local/sbin/opensips[622409]:
>>>             ERROR:stir_shaken:w_stir_verify: failed to validate
>>>             Destination number (3310200123456789)
>>>
>>>             _*My configuration:*_
>>>             # ----------------- module  stir_shaken ---------------
>>>             loadmodule "stir_shaken.so"
>>>             #----------- stir_shaken params -----------------
>>>             modparam("stir_shaken", "ca_list",
>>>             "/usr/local/etc/opensips/man_ca.pem")
>>>             modparam("stir_shaken", "require_date_hdr", 0)
>>>             modparam("stir_shaken", "verify_date_freshness", 60)
>>>
>>>             According to the doc e164_strict_mode is disabled by
>>>             default, so I don't know why it doesn't work.
>>>
>>>             _*source of code: *_
>>>                     if (_is_e164(num, e164_strict_mode) == -1) {
>>>                             LM_GEN(log_lev, "number is not in E.164
>>>             format: %.*s\n", num->len, num->s);
>>>                             return -1;
>>>                     }
>>>
>>>
>>>             Do you have any help for me please ? I have to validate
>>>             this format of dest number.
>>>
>>>             Thanks in advance
>>>
>>>
>>>             _______________________________________________
>>>             Users mailing list
>>>             Users at lists.opensips.org
>>>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>             -- 
>>             Best Regards:
>>             Marcin Groszek
>>             Business Phone Service
>>             https://www.voipplus.net
>>
>>             _______________________________________________
>>             Users mailing list
>>             Users at lists.opensips.org
>>             http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>             -- 
>             Best Regards:
>             Marcin Groszek
>             Business Phone Service
>             https://www.voipplus.net
>
>             _______________________________________________
>             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 mickael at winlux.fr  Thu Sep  7 09:33:47 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Thu, 7 Sep 2023 11:33:47 +0200
Subject: [OpenSIPS-Users] Stir ans Shaken - number is not in E.164 format
In-Reply-To: <a4fe9736-462b-03f0-0ca3-79a2eaf2fcbc@democon.be>
References: <CA+fRVSU2PhdLiz0wYqvuODmtci5Bx0O_u4AJqyf9SSG5Ux3eHQ@mail.gmail.com>
 <5046d2bc-2200-2655-c42f-69ac290de066@voipplus.net>
 <9d184b36-d422-1c05-f662-339203f40bcf@voipplus.net>
 <CA+fRVSWKn=XeR7RtmmUKh-T7pb=sm4584-OCtQfirFWxfshmrA@mail.gmail.com>
 <CAFGRPVrmzwhbPZHjyBbHwQPPQhNX=5XSLGtvgGczpKk8M4EADA@mail.gmail.com>
 <CAPwC5wy1h=ya3q48m5ucNJtfyWdu7nJGVeQrJ9ts=YHeZnedkw@mail.gmail.com>
 <a4fe9736-462b-03f0-0ca3-79a2eaf2fcbc@democon.be>
Message-ID: <CA+fRVSXWJfET2qBsARLpSqnL_7X12Z2Opyxs5FCLRxjxRLCCAQ@mail.gmail.com>

Hi all,
thanks Brett for this link, very interesting.
I received an official answer, and yes in France it's possible to sign
without E164
my explanation (in french here):
https://www.mail-archive.com/frnog at frnog.org/msg73317.html

I patched stir and shaken module and that works.

++

Le jeu. 7 sept. 2023 à 09:45, johan <johan at democon.be> a écrit :

> I think that having this on a global scale will result in a mumbo jumbo of
> standards and implementations.  Best thing for Europe would be that this
> would be tackled on EU scale, but it don't see this happening (e.g. what
> France does, is for sure not okay for Germany and so on).   Anyway, I will
> be on that call too, as I would like to hear their view on this complicated
> item.
> On 6/09/2023 19:21, Brett Nemeroff wrote:
>
> Buckle up! Things are going to get interesting. This is just one tiny
> little example...
>
> We have a lot of international issues when it comes to building trust and
> call authentication internationally. For example, consider that STIR/SHAKEN
> as we know it is built upon the US policy and governance authority and as
> such certificates come from US STI-CA. What will happen internationally
> with certificates and trust? Will PASSporTs all look the same? How do we
> trust calls that cross borders?
>
> PASSporT formatting is the tip of the iceberg. However for France calling,
> they have their own set of standards governed by MAN. OpenSIPs modules
> should certainly have a way to support them, but I would be surprised if
> they do out of the box. Given how new all of this is, I'd expect we'd need
> the help of someone like Mickael to help understand where the module is
> lacking for France standards.
>
> For those who are interested, i3Forum is having a talk tomorrow morning
> (US) on international calling and trust which may cover some of these
> issues. I am not affiliated with that group, but I'll be joining the
> webinar to gather information. I'm passing the information on to you guys.
>
> https://i3forum.org/one-consortium/
>
> -Brett
>
>
> On Wed, Sep 6, 2023 at 7:31 AM David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Is ST/SH being used other than the US? AFAIK it only applies to US
>> numbers, thus 10 digits, no?
>>
>> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert <mickael at winlux.fr> wrote:
>>
>>> yep I found...
>>>
>>> if (end - start < 2 || end - start > 15)
>>> return -1;
>>>
>>> I have to modify this code.
>>> I will propose a PR.
>>>
>>> Thanks a lot
>>> ++
>>>
>>> Le mer. 6 sept. 2023 à 14:25, Marcin Groszek <marcin at voipplus.net> a
>>> écrit :
>>>
>>>> Correction : maximum of 15 digits .
>>>> On 9/6/2023 7:21 AM, Marcin Groszek wrote:
>>>>
>>>> Your number is to long
>>>>
>>>> E.164 is + [1-9]  and  {1-14} digits for total of 15 digits NOT
>>>> starting with 0
>>>> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>>>>
>>>> Hi all,
>>>> I have an issue, when I verify a call with no E164 format (dest:
>>>> +3310200123456789)
>>>>
>>>> *logs:*
>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
>>>> 3310200123456789
>>>> Sep  6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
>>>> ERROR:stir_shaken:w_stir_verify: failed to validate Destination number
>>>> (3310200123456789)
>>>>
>>>> *My configuration:*
>>>> # ----------------- module  stir_shaken ---------------
>>>> loadmodule "stir_shaken.so"
>>>> #----------- stir_shaken params -----------------
>>>> modparam("stir_shaken", "ca_list", "/usr/local/etc/opensips/man_ca.pem")
>>>> modparam("stir_shaken", "require_date_hdr", 0)
>>>> modparam("stir_shaken", "verify_date_freshness", 60)
>>>>
>>>> According to the doc e164_strict_mode is disabled by default, so I
>>>> don't know why it doesn't work.
>>>>
>>>> *source of code: *
>>>>         if (_is_e164(num, e164_strict_mode) == -1) {
>>>>                 LM_GEN(log_lev, "number is not in E.164 format:
>>>> %.*s\n", num->len, num->s);
>>>>                 return -1;
>>>>         }
>>>>
>>>>
>>>> Do you have any help for me please ? I have to validate this format of
>>>> dest number.
>>>>
>>>> Thanks in advance
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>>
>>>> _______________________________________________
>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>>
>>>> --
>>>> Best Regards:
>>>> Marcin Groszek
>>>> Business Phone Servicehttps://www.voipplus.net
>>>>
>>>> _______________________________________________
>>>> 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 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 prathibhab.tvm at gmail.com  Thu Sep  7 09:48:49 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Thu, 7 Sep 2023 15:18:49 +0530
Subject: [OpenSIPS-Users] users not getting registered
Message-ID: <CAJh_-cAM42UYHgx3u5aKrtpzEj_ffCTthbef-WDSt_wfhfzAUw@mail.gmail.com>

I've configured opensips and added two users in opensips control panel.
When checking the status of opensips service through

systemctl status opensips, I get the following message

WARNING:dialplan:dp_load_db: no data in the db



-- 
Regards,
B.Prathibha
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From iamhalje at gmail.com  Thu Sep  7 11:49:10 2023
From: iamhalje at gmail.com (Dmitry Ponomaryov)
Date: Thu, 7 Sep 2023 16:49:10 +0500
Subject: [OpenSIPS-Users] Increased macro MAX_BRANCHES and behavior of tm
	module
Message-ID: <CAJUEF2uhxMBW-vLLarr1Demo7ypHnja5YQt3BoWhBAZ9Wd=ZDQ@mail.gmail.com>

Have a nice day, everyone

Question is to increase the value of MAX_BRANCHES[1] to 32, for example,
which will go beyond the boundaries of the source code in tm.c, namely
condition[2], which clearly shows that only 30 is possible, how critical it
is to change the condition in tm.c, in order to get the need value? [1]
https://github.com/OpenSIPS/opensips/blob/master/config.h#L169 [2]
https://github.com/OpenSIPS/opensips/blob/master/modules/tm/tm.c#L817-L821
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From prathibhab.tvm at gmail.com  Thu Sep  7 11:54:35 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Thu, 7 Sep 2023 17:24:35 +0530
Subject: [OpenSIPS-Users] opensips user registration failed
Message-ID: <CAJh_-cAat8-cfNcj7N93cUKHB-i0B3ocTUWDrD1qcp1ucRRFgg@mail.gmail.com>

I've resolved the issues in the opensips configuration file and the
"systemctl status opensips" shows no error. However I'm not able to
register a user through zoiper(soft phone). How to identify the cause for
this?

*systemctl status opensips*
opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261) server
     Loaded: loaded (/lib/systemd/system/opensips.service; enabled; vendor
preset: enabled)
     Active: active (running) since Thu 2023-09-07 11:36:15 UTC; 4s ago
       Docs: man:opensips
    Process: 245256 ExecStart=/usr/sbin/opensips -P
/run/opensips/opensips.pid -f /etc/opensips/opensips.cfg.orig4 -m $S_MEMORY
-M $P_MEMORY $OPTIONS (code=exited, status=0/SUCCESS)
   Main PID: 245260 (opensips)
      Tasks: 19 (limit: 1121)
     Memory: 28.5M
        CPU: 268ms
     CGroup: /system.slice/opensips.service
             ├─245260 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245261 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245262 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245263 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245264 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245265 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245266 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245267 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245268 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245269 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245270 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245271 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245272 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245273 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245274 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245275 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245276 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             ├─245277 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4
             └─245278 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
/etc/opensips/opensips.cfg.orig4 -m 64 -M 4

Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245273]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found, support
for it enabled
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245276]:
INFO:rtpproxy:connect_rtpproxies: created to 146
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245274]:
INFO:rtpproxy:connect_rtpproxies: created to 146
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245277]:
INFO:rtpproxy:connect_rtpproxies: created to 146
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245275]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found, support
for it enabled
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245276]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found, support
for it enabled
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245274]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found, support
for it enabled
Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245277]:
INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found, support
for it enabled
Sep 07 11:36:15 ip-172-31-34-24 opensips[245256]: INFO:core:daemonize:
pre-daemon process exiting with 0
Sep 07 11:36:15 ip-172-31-34-24 systemd[1]: Started OpenSIPS is a very fast
and flexible SIP (RFC3261) server.


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Thu Sep  7 11:57:47 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Thu, 7 Sep 2023 17:27:47 +0530
Subject: [OpenSIPS-Users] opensips user registration failed
In-Reply-To: <CAJh_-cAat8-cfNcj7N93cUKHB-i0B3ocTUWDrD1qcp1ucRRFgg@mail.gmail.com>
References: <CAJh_-cAat8-cfNcj7N93cUKHB-i0B3ocTUWDrD1qcp1ucRRFgg@mail.gmail.com>
Message-ID: <CAJh_-cCASyBV16HdVn8znrTyWo+E6LqjTH8Fr57TvK+M7gUTSw@mail.gmail.com>

opensips.cfg file is also attached.

On Thu, 7 Sept 2023 at 17:24, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I've resolved the issues in the opensips configuration file and the
> "systemctl status opensips" shows no error. However I'm not able to
> register a user through zoiper(soft phone). How to identify the cause for
> this?
>
> *systemctl status opensips*
> opensips.service - OpenSIPS is a very fast and flexible SIP (RFC3261)
> server
>      Loaded: loaded (/lib/systemd/system/opensips.service; enabled; vendor
> preset: enabled)
>      Active: active (running) since Thu 2023-09-07 11:36:15 UTC; 4s ago
>        Docs: man:opensips
>     Process: 245256 ExecStart=/usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg.orig4 -m $S_MEMORY
> -M $P_MEMORY $OPTIONS (code=exited, status=0/SUCCESS)
>    Main PID: 245260 (opensips)
>       Tasks: 19 (limit: 1121)
>      Memory: 28.5M
>         CPU: 268ms
>      CGroup: /system.slice/opensips.service
>              ├─245260 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245261 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245262 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245263 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245264 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245265 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245266 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245267 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245268 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245269 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245270 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245271 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245272 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245273 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245274 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245275 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245276 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              ├─245277 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>              └─245278 /usr/sbin/opensips -P /run/opensips/opensips.pid -f
> /etc/opensips/opensips.cfg.orig4 -m 64 -M 4
>
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245273]:
> INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found,
> support for it enabled
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245276]:
> INFO:rtpproxy:connect_rtpproxies: created to 146
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245274]:
> INFO:rtpproxy:connect_rtpproxies: created to 146
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245277]:
> INFO:rtpproxy:connect_rtpproxies: created to 146
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245275]:
> INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found,
> support for it enabled
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245276]:
> INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found,
> support for it enabled
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245274]:
> INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found,
> support for it enabled
> Sep 07 11:36:15 ip-172-31-34-24 /usr/sbin/opensips[245277]:
> INFO:rtpproxy:rtpp_test: rtp proxy <udp:172.31.34.24:60000> found,
> support for it enabled
> Sep 07 11:36:15 ip-172-31-34-24 opensips[245256]: INFO:core:daemonize:
> pre-daemon process exiting with 0
> Sep 07 11:36:15 ip-172-31-34-24 systemd[1]: Started OpenSIPS is a very
> fast and flexible SIP (RFC3261) server.
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From alberto.rinaudo at gmail.com  Fri Sep  8 12:41:31 2023
From: alberto.rinaudo at gmail.com (Alberto)
Date: Fri, 8 Sep 2023 13:41:31 +0100
Subject: [OpenSIPS-Users] sip info dtmf
Message-ID: <CANoA6_LO3AcQV5gBt-EN4H7zc5NMrC1421-5w4=JfKy3j+9Jmg@mail.gmail.com>

Hi,
I'm using opensips 3.2 and can't find any way to get this working.
Can someone help me complete this code?
I had a look at textops, but I don't see any function that can extract a
regex matched group.
Thanks

if (is_method("INFO") && $hdr(Content-Type) == "application/dtmf-relay") {
  $var(body) = $rb(application/dtmf-relay);
  if ($var(body) =~ "Signal=([0-9]+)") {
     $var(dtmf) = ???;
     xlog("L_NOTICE", "$$var(dtmf): $var(dtmf)\n");
   }
}
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From venefax at gmail.com  Sun Sep 10 05:33:37 2023
From: venefax at gmail.com (Saint Michael)
Date: Sun, 10 Sep 2023 01:33:37 -0400
Subject: [OpenSIPS-Users] upgrade failed
Message-ID: <CAC9cSOBTr2z9kY5GJ=eqG1PeaNzBH2fO32A4JMNF7DQUniUORw@mail.gmail.com>

I attempted an upgrade to version 3.4 from 3.1 and It failed
ERROR:core:db_check_table_version: invalid version 8 for table dispatcher
found, expected 9
ERROR:dispatcher:mod_init: failed to init database support
ERROR:core:init_mod: failed to initialize module dispatcher
ERROR:core:main: error while initializing modules

how do I upgrade the dispatcher table from 8 version 8 to version 9?
The script is fine.
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From social at bohboh.info  Sun Sep 10 07:02:39 2023
From: social at bohboh.info (Social Boh)
Date: Sun, 10 Sep 2023 02:02:39 -0500
Subject: [OpenSIPS-Users] upgrade failed
In-Reply-To: <CAC9cSOBTr2z9kY5GJ=eqG1PeaNzBH2fO32A4JMNF7DQUniUORw@mail.gmail.com>
References: <CAC9cSOBTr2z9kY5GJ=eqG1PeaNzBH2fO32A4JMNF7DQUniUORw@mail.gmail.com>
Message-ID: <edbd3677-c1d2-e0a7-49b7-7595af38162d@bohboh.info>

mysql -u root -p

use opensips

update version set table_version='9' where table_name='dispatcher';

quit

---
I'm SoCIaL, MayBe

El 10/09/2023 a las 12:33 a. m., Saint Michael escribió:
> I attempted an upgrade to version 3.4 from 3.1 and It failed
> ERROR:core:db_check_table_version: invalid version 8 for table 
> dispatcher found, expected 9
> ERROR:dispatcher:mod_init: failed to init database support
> ERROR:core:init_mod: failed to initialize module dispatcher
> ERROR:core:main: error while initializing modules
>
> how do I upgrade the dispatcher table from 8 version 8 to version 9?
> The script is fine.
>
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From prathibhab.tvm at gmail.com  Mon Sep 11 08:10:01 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 11 Sep 2023 13:40:01 +0530
Subject: [OpenSIPS-Users] Error while loading private key
Message-ID: <CAJh_-cCk3EfwQhULtVXwfQQdm+vPkzeXfknoTtdHndGQGhyN3w@mail.gmail.com>

ERROR:tls_wolfssl:load_private_key: key
'/etc/opensips/tls/user/user-privkey.pem' does not match the public key of
the certificate
Sep 11 08:05:11 ip-172-31-34-24 /usr/sbin/opensips[286044]:
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Mon Sep 11 08:10:46 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 11 Sep 2023 13:40:46 +0530
Subject: [OpenSIPS-Users] Error while loading private key
In-Reply-To: <CAJh_-cCk3EfwQhULtVXwfQQdm+vPkzeXfknoTtdHndGQGhyN3w@mail.gmail.com>
References: <CAJh_-cCk3EfwQhULtVXwfQQdm+vPkzeXfknoTtdHndGQGhyN3w@mail.gmail.com>
Message-ID: <CAJh_-cDwYqkNED4z=Wcj674YqvstAc=Jc_59upV2zn4HDF7buw@mail.gmail.com>

I'm using the correct certificate. Still I'm getting the above error
message.

On Mon, 11 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> ERROR:tls_wolfssl:load_private_key: key
> '/etc/opensips/tls/user/user-privkey.pem' does not match the public key of
> the certificate
> Sep 11 08:05:11 ip-172-31-34-24 /usr/sbin/opensips[286044]:
> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Mon Sep 11 08:38:55 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 11 Sep 2023 14:08:55 +0530
Subject: [OpenSIPS-Users] Error while loading private key
In-Reply-To: <CAJh_-cDwYqkNED4z=Wcj674YqvstAc=Jc_59upV2zn4HDF7buw@mail.gmail.com>
References: <CAJh_-cCk3EfwQhULtVXwfQQdm+vPkzeXfknoTtdHndGQGhyN3w@mail.gmail.com>
 <CAJh_-cDwYqkNED4z=Wcj674YqvstAc=Jc_59upV2zn4HDF7buw@mail.gmail.com>
Message-ID: <CAJh_-cBFKhEM8nZH2_gCVmJ+NSqYJPy_zW1J4gNDHqMjqHz8Nw@mail.gmail.com>

with openssl, I'm getting
ERROR:tls_openssl:load_private_key: unable to load private key file
'/etc/opensips/tls/user/user-privkey.pem'. #012Retry (2 left) (check
password case)
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
ERROR:tls_openssl:load_private_key: unable to load private key file
'/etc/opensips/tls/user/user-privkey.pem'. #012Retry (1 left) (check
password case)
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
ERROR:tls_openssl:load_private_key: unable to load private key file
'/etc/opensips/tls/user/user-privkey.pem'. #012Retry (0 left) (check
password case)
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:05800074:x509
certificate routines::key values mismatch
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]: message
repeated 2 times: [ ERROR:tls_openssl:tls_print_errstack: TLS errstack:
error:05800074:x509 certificate routines::key values mismatch]
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
ERROR:tls_openssl:load_private_key: unable to load private key file
'/etc/opensips/tls/user/user-privkey.pem'
Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'

On Mon, 11 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I'm using the correct certificate. Still I'm getting the above error
> message.
>
> On Mon, 11 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> ERROR:tls_wolfssl:load_private_key: key
>> '/etc/opensips/tls/user/user-privkey.pem' does not match the public key of
>> the certificate
>> Sep 11 08:05:11 ip-172-31-34-24 /usr/sbin/opensips[286044]:
>> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Mon Sep 11 08:57:23 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 11 Sep 2023 14:27:23 +0530
Subject: [OpenSIPS-Users] Error while loading private key
In-Reply-To: <CAJh_-cBFKhEM8nZH2_gCVmJ+NSqYJPy_zW1J4gNDHqMjqHz8Nw@mail.gmail.com>
References: <CAJh_-cCk3EfwQhULtVXwfQQdm+vPkzeXfknoTtdHndGQGhyN3w@mail.gmail.com>
 <CAJh_-cDwYqkNED4z=Wcj674YqvstAc=Jc_59upV2zn4HDF7buw@mail.gmail.com>
 <CAJh_-cBFKhEM8nZH2_gCVmJ+NSqYJPy_zW1J4gNDHqMjqHz8Nw@mail.gmail.com>
Message-ID: <CAJh_-cBpZSfG9OmYAOqa9gfSOSbL3LvoVojbHnhyJHxuc=d0fQ@mail.gmail.com>

ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
14.139.183.221:61764 failed to accept
Sep 11 08:56:54 ip-172-31-34-24 /usr/sbin/opensips[286807]:
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A0000BF:SSL
routines::no protocols available
Sep 11 08:56:54 ip-172-31-34-24 /usr/sbin/opensips[286807]:
ERROR:proto_wss:wss_read_req: cannot fix read connection

On Mon, 11 Sept 2023 at 14:08, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> with openssl, I'm getting
> ERROR:tls_openssl:load_private_key: unable to load private key file
> '/etc/opensips/tls/user/user-privkey.pem'. #012Retry (2 left) (check
> password case)
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
> ERROR:tls_openssl:load_private_key: unable to load private key file
> '/etc/opensips/tls/user/user-privkey.pem'. #012Retry (1 left) (check
> password case)
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
> ERROR:tls_openssl:load_private_key: unable to load private key file
> '/etc/opensips/tls/user/user-privkey.pem'. #012Retry (0 left) (check
> password case)
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
> ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:05800074:x509
> certificate routines::key values mismatch
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]: message
> repeated 2 times: [ ERROR:tls_openssl:tls_print_errstack: TLS errstack:
> error:05800074:x509 certificate routines::key values mismatch]
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
> ERROR:tls_openssl:load_private_key: unable to load private key file
> '/etc/opensips/tls/user/user-privkey.pem'
> Sep 11 08:37:59 ip-172-31-34-24 /usr/sbin/opensips[286530]:
> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'
>
> On Mon, 11 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> I'm using the correct certificate. Still I'm getting the above error
>> message.
>>
>> On Mon, 11 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com>
>> wrote:
>>
>>> ERROR:tls_wolfssl:load_private_key: key
>>> '/etc/opensips/tls/user/user-privkey.pem' does not match the public key of
>>> the certificate
>>> Sep 11 08:05:11 ip-172-31-34-24 /usr/sbin/opensips[286044]:
>>> ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'bp3.erss.in'
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Mon Sep 11 10:04:52 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Mon, 11 Sep 2023 15:34:52 +0530
Subject: [OpenSIPS-Users] cannot fix read connection
Message-ID: <CAJh_-cD0atGGjQojAn2b43Fg3Wt0-6nedGWovZ8fR=6Y3dv1aQ@mail.gmail.com>

ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
14.139.183.221:59966 failed to accept
Sep 11 10:02:51 ip-172-31-34-24 /usr/sbin/opensips[287604]:
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A0000BF:SSL
routines::no protocols available
Sep 11 10:02:51 ip-172-31-34-24 /usr/sbin/opensips[287604]:
ERROR:proto_wss:wss_read_req: cannot fix read connection

How to resolve the above issue?

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Tue Sep 12 04:47:36 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 12 Sep 2023 10:17:36 +0530
Subject: [OpenSIPS-Users] Request timed out
Message-ID: <CAJh_-cABtqiHb9TH-C=uQxh1LTBGdFf9Nc7gmJcdwXwKjj3=5w@mail.gmail.com>

I've setup opensips. Websocket is getting connected. But when I try to
register user, I get the following message:

Registration Failed: Request Timeout

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Thu Sep 14 12:53:35 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Thu, 14 Sep 2023 12:53:35 +0000
Subject: [OpenSIPS-Users] Configuration file
Message-ID: <KL1PR0601MB4004AF6B0D0B49B3A188BB05FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>

Can anyone share the opensips conf file with rtpengine?

Sent from Outlook for Android<https://aka.ms/AAb9ysg>
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From 9to1url at gmail.com  Thu Sep 14 13:12:19 2023
From: 9to1url at gmail.com (Nine to one)
Date: Thu, 14 Sep 2023 09:12:19 -0400
Subject: [OpenSIPS-Users] Configure file for opensips 3.4
Message-ID: <CAOPUfWXgu3q2_tS-J2Ra-0WZHm0KR6UccbrR1fnJaW3bNM7FCA@mail.gmail.com>

Can anyone share the opensips conf file with Freeswitch and rtpengine?
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From prathibhab.tvm at gmail.com  Thu Sep 14 14:19:31 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Thu, 14 Sep 2023 14:19:31 +0000
Subject: [OpenSIPS-Users] Configuration file
In-Reply-To: <KL1PR0601MB4004AF6B0D0B49B3A188BB05FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
References: <KL1PR0601MB4004AF6B0D0B49B3A188BB05FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
Message-ID: <KL1PR0601MB4004DBD66B561FA9E5717DE4FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>

I need the conf file for opensips 3.3

Sent from Outlook for Android<https://aka.ms/AAb9ysg>
________________________________
From: Prathibha B <prathibhab.tvm at gmail.com>
Sent: Thursday, September 14, 2023 6:23:35 PM
To: users at lists.opensips.org <Users at lists.opensips.org>
Subject: Configuration file

Can anyone share the opensips conf file with rtpengine?

Sent from Outlook for Android<https://aka.ms/AAb9ysg>
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From callum.guy at x-on.co.uk  Thu Sep 14 18:20:24 2023
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Thu, 14 Sep 2023 19:20:24 +0100
Subject: [OpenSIPS-Users] Configuration file
In-Reply-To: <KL1PR0601MB4004DBD66B561FA9E5717DE4FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
References: <KL1PR0601MB4004AF6B0D0B49B3A188BB05FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
 <KL1PR0601MB4004DBD66B561FA9E5717DE4FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
Message-ID: <CAFjCFzk=2OizV4OnLDFWVDn0y1nPpk7sGw5ab=dBTWmAM84hvg@mail.gmail.com>

The idea is that you create your own based on your unique
infrastructure requirements.

Learn you must.

https://www.opensips.org/Documentation/Manual-3-3


On Thu, 14 Sept 2023 at 15:21, Prathibha B <prathibhab.tvm at gmail.com> wrote:
>
> I need the conf file for opensips 3.3
>
> Sent from Outlook for Android
> ________________________________
> From: Prathibha B <prathibhab.tvm at gmail.com>
> Sent: Thursday, September 14, 2023 6:23:35 PM
> To: users at lists.opensips.org <Users at lists.opensips.org>
> Subject: Configuration file
>
> Can anyone share the opensips conf file with rtpengine?
>
> Sent from Outlook for Android
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 






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From Johan at democon.be  Thu Sep 14 18:24:46 2023
From: Johan at democon.be (Johan De Clercq)
Date: Thu, 14 Sep 2023 20:24:46 +0200
Subject: [OpenSIPS-Users] Configuration file
In-Reply-To: <CAFjCFzk=2OizV4OnLDFWVDn0y1nPpk7sGw5ab=dBTWmAM84hvg@mail.gmail.com>
References: <KL1PR0601MB4004AF6B0D0B49B3A188BB05FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
 <KL1PR0601MB4004DBD66B561FA9E5717DE4FCF7A@KL1PR0601MB4004.apcprd06.prod.outlook.com>
 <CAFjCFzk=2OizV4OnLDFWVDn0y1nPpk7sGw5ab=dBTWmAM84hvg@mail.gmail.com>
Message-ID: <CAEVWGm-56ypMzU2rsqHUN_BUcHA1+7p8xU6wi9KtJ_2Z-WwDgg@mail.gmail.com>

Indeed. Everything starts with reading documentation.

On Thu, 14 Sept 2023, 20:23 Callum Guy, <callum.guy at x-on.co.uk> wrote:

> The idea is that you create your own based on your unique
> infrastructure requirements.
>
> Learn you must.
>
> https://www.opensips.org/Documentation/Manual-3-3
>
>
> On Thu, 14 Sept 2023 at 15:21, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
> >
> > I need the conf file for opensips 3.3
> >
> > Sent from Outlook for Android
> > ________________________________
> > From: Prathibha B <prathibhab.tvm at gmail.com>
> > Sent: Thursday, September 14, 2023 6:23:35 PM
> > To: users at lists.opensips.org <Users at lists.opensips.org>
> > Subject: Configuration file
> >
> > Can anyone share the opensips conf file with rtpengine?
> >
> > Sent from Outlook for Android
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
>
>
>
>
>
>
> *0333 332 0000  |  x-on.co.uk <https://www.x-on.co.uk>  |   **
> <https://www.linkedin.com/company/x-on>   <https://www.facebook.com/XonTel>
>
>   <https://twitter.com/xonuk> **  |  **Practice Index Reviews
> <https://practiceindex.co.uk/gp/x-on>*
>
> *Our new office address: 22 Riduna
> Park, Melton IP12 1QT.*
>
> X-on
> is a trading name of X-on Health Ltd a
> limited company registered in
> England and Wales.
>
> Registered Office : Glebe
> Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD.
> Company Registration No. 2578478.
>
> The information in this e-mail is
> confidential and for use by the addressee(s)
> only. If you are not the
> intended recipient, please notify X-on immediately on +44(0)333 332 0000
> and delete the
> message from your computer. If you are not a named addressee
> you must not use,
> disclose, disseminate, distribute, copy, print or reply
> to this email. Views
> or opinions expressed by an individual
> within this
> email may not necessarily
> reflect the views of X-on or its associated
> companies. Although X-on routinely
> screens for viruses, addressees should
> scan this email and any attachments
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> viruses. X-on makes no
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From 9to1url at gmail.com  Thu Sep 14 20:40:51 2023
From: 9to1url at gmail.com (Nine to one)
Date: Thu, 14 Sep 2023 16:40:51 -0400
Subject: [OpenSIPS-Users] Configure file for opensips 3.4
In-Reply-To: <CAOPUfWXgu3q2_tS-J2Ra-0WZHm0KR6UccbrR1fnJaW3bNM7FCA@mail.gmail.com>
References: <CAOPUfWXgu3q2_tS-J2Ra-0WZHm0KR6UccbrR1fnJaW3bNM7FCA@mail.gmail.com>
Message-ID: <CAOPUfWX+drs0-U3U28d_A6Agc8xKGdCKmdje7qPt2U17pOC3QQ@mail.gmail.com>

Get idea from another thread: Reading documents :-)
Did that.

Nine to one <9to1url at gmail.com> 於 2023年9月14日 週四 上午9:12寫道：

> Can anyone share the opensips conf file with Freeswitch and rtpengine?
>
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From 9to1url at gmail.com  Thu Sep 14 20:46:26 2023
From: 9to1url at gmail.com (Nine to one)
Date: Thu, 14 Sep 2023 16:46:26 -0400
Subject: [OpenSIPS-Users] Configure file for opensips 3.4
In-Reply-To: <CAOPUfWX+drs0-U3U28d_A6Agc8xKGdCKmdje7qPt2U17pOC3QQ@mail.gmail.com>
References: <CAOPUfWXgu3q2_tS-J2Ra-0WZHm0KR6UccbrR1fnJaW3bNM7FCA@mail.gmail.com>
 <CAOPUfWX+drs0-U3U28d_A6Agc8xKGdCKmdje7qPt2U17pOC3QQ@mail.gmail.com>
Message-ID: <CAOPUfWV6uj=Tv6kW34LNtQ8ZW5AbAGa3P6K3R4fatPJC2KD8Gg@mail.gmail.com>

After reading many docs, googled, chatgpt, I barely have a working
configuration with only REGISTER working , but INVITE not working:

``` opensips.cfg
#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team at opensips-solutions.com>
#
# This script was generated via "make menuconfig", from
#   the "Residential" scenario.
# You can enable / disable more features / functionalities by
#   re-generating the scenario with different options.#
#
# Please refer to the Core CookBook at:
#      https://opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

/* uncomment the following lines to enable debugging */
#debug_mode=yes

### default log level is 3, change to 4 for debugging
log_level=4
xlog_level=4
stderror_enabled=yes
syslog_enabled=yes
syslog_facility=LOG_LOCAL0

udp_workers=4

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes


#socket=udp:127.0.0.1:5060   # CUSTOMIZE ME
socket=udp:192.168.113.144:5060
#listen=udp:192.168.113.144:5060



####### Modules Section ########

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"


### add for freeswitch
loadmodule "db_postgres.so"


#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

### add for freeswitch
loadmodule "dialog.so"
modparam("dialog", "db_mode", 1)
modparam("dialog", "db_url", "postgres://
opensips:opensipsrw at 192.168.113.145/opensips")


#### MAX ForWarD module
loadmodule "maxfwd.so"

### add for freeswitch
loadmodule "textops.so"


#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/run/opensips/opensips_fifo")
# not working either, modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

### add for freeswitch
loadmodule "dispatcher.so"
modparam("dispatcher", "db_url", "postgres://
opensips:opensipsrw at 192.168.113.145/opensips")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 5)
modparam("dispatcher", "ds_probing_threshhold", 2)
modparam("dispatcher", "ds_probing_mode", 1)

### add for freeswitch
loadmodule "load_balancer.so"
modparam("load_balancer", "db_url", "postgres://
opensips:opensipsrw at 192.168.113.145/opensips")
modparam("load_balancer", "probing_method", "OPTIONS")
modparam("load_balancer", "probing_interval", 5)



#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-no-db")

#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure to enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)

loadmodule "proto_udp.so"


#### gjw add for rtpengine
loadmodule "rtpengine.so"
# single rtproxy
modparam("rtpengine", "rtpengine_sock", "udp:192.168.113.143:2223")
# multiple rtproxies for LB
#modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221
udp:localhost:12222")
# multiple sets of multiple rtproxies
#modparam("rtpengine", "rtpengine_sock", "1 == udp:localhost:12221
udp:localhost:12222")
#modparam("rtpengine", "rtpengine_sock", "2 == udp:localhost:12225")

### add for freeswitch



####### Routing Logic ########

# main request routing logic

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");
#                # Route to FreeSWITCH
#                $du = "sip:192.168.113.142:5060";
#                forward();

if (!load_balance(1,"pstn","1")) {
    log("sreekanth invite method\n");
    send_reply(503,"Service Unavailable");
    exit;
}

}


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")) {
# # store the registration and generate a SIP reply
# if (!save("location"))
# xlog("failed to register AoR $tu\n");
#
# exit;


if (!ds_select_dst(1, 4)) {
    send_reply(503,"Service Unavailable");
    exit;
}
}

# if ($rU==NULL) {
# # request with no Username in RURI
# send_reply(484,"Address Incomplete");
# exit;
# }

# do lookup with method filtering
if (!lookup("location","method-filtering")) {
t_reply(404, "Not Found");
exit;
}

# when routing via usrloc, log the missed calls also
do_accounting("log","missed");
route(relay);
}


route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
t_on_branch("per_branch_ops");
t_on_reply("handle_nat");
t_on_failure("missed_call");
}

if (!t_relay()) {
send_reply(500,"Internal Error");
}
exit;
}




branch_route[per_branch_ops] {
xlog("new branch at $ru\n");
}


onreply_route[handle_nat] {
xlog("incoming reply\n");
}


failure_route[missed_call] {
if (t_was_cancelled()) {
exit;
}

# uncomment the following lines if you want to block client
# redirect based on 3xx replies.
##if (t_check_status("3[0-9][0-9]")) {
##t_reply(404,"Not found");
## exit;
##}


}

```

Any idea why INVITE not working? :-)


Nine to one <9to1url at gmail.com> 於 2023年9月14日 週四 下午4:40寫道：

> Get idea from another thread: Reading documents :-)
> Did that.
>
> Nine to one <9to1url at gmail.com> 於 2023年9月14日 週四 上午9:12寫道：
>
>> Can anyone share the opensips conf file with Freeswitch and rtpengine?
>>
>
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From social at bohboh.info  Thu Sep 14 21:20:26 2023
From: social at bohboh.info (Social Boh)
Date: Thu, 14 Sep 2023 16:20:26 -0500
Subject: [OpenSIPS-Users] Configure file for opensips 3.4
In-Reply-To: <CAOPUfWV6uj=Tv6kW34LNtQ8ZW5AbAGa3P6K3R4fatPJC2KD8Gg@mail.gmail.com>
References: <CAOPUfWXgu3q2_tS-J2Ra-0WZHm0KR6UccbrR1fnJaW3bNM7FCA@mail.gmail.com>
 <CAOPUfWX+drs0-U3U28d_A6Agc8xKGdCKmdje7qPt2U17pOC3QQ@mail.gmail.com>
 <CAOPUfWV6uj=Tv6kW34LNtQ8ZW5AbAGa3P6K3R4fatPJC2KD8Gg@mail.gmail.com>
Message-ID: <7f15b860-2dd6-3059-928b-0e1e7c919d0c@bohboh.info>

after reading nothing, after debugging nothing you still creating noise 
in this list. Maybe moderators have to think todo something with you and 
your very high annoying messages

---
I'm SoCIaL, MayBe

El 14/09/2023 a las 3:46 p. m., Nine to one escribió:
> After reading many docs, googled, chatgpt, I barely have a working 
> configuration with only REGISTER working , but INVITE not working:
>
> ``` opensips.cfg
> #
> # OpenSIPS residential configuration script
> #     by OpenSIPS Solutions <team at opensips-solutions.com>
> #
> # This script was generated via "make menuconfig", from
> #   the "Residential" scenario.
> # You can enable / disable more features / functionalities by
> #   re-generating the scenario with different options.#
> #
> # Please refer to the Core CookBook at:
> # https://opensips.org/Resources/DocsCookbooks
> # for a explanation of possible statements, functions and parameters.
> #
>
>
> ####### Global Parameters #########
>
> /* uncomment the following lines to enable debugging */
> #debug_mode=yes
>
> ### default log level is 3, change to 4 for debugging
> log_level=4
> xlog_level=4
> stderror_enabled=yes
> syslog_enabled=yes
> syslog_facility=LOG_LOCAL0
>
> udp_workers=4
>
> /* uncomment the next line to enable the auto temporary blacklisting of
>    not available destinations (default disabled) */
> #disable_dns_blacklist=no
>
> /* uncomment the next line to enable IPv6 lookup after IPv4 dns
>    lookup failures (default disabled) */
> #dns_try_ipv6=yes
>
>
> #socket=udp:127.0.0.1:5060 <http://127.0.0.1:5060>   # CUSTOMIZE ME
> socket=udp:192.168.113.144:5060 <http://192.168.113.144:5060>
> #listen=udp:192.168.113.144:5060 <http://192.168.113.144:5060>
>
>
>
> ####### Modules Section ########
>
> #set module path
> mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"
>
>
> ### add for freeswitch
> loadmodule "db_postgres.so"
>
>
> #### SIGNALING module
> loadmodule "signaling.so"
>
> #### StateLess module
> loadmodule "sl.so"
>
> #### Transaction Module
> loadmodule "tm.so"
> modparam("tm", "fr_timeout", 5)
> modparam("tm", "fr_inv_timeout", 30)
> modparam("tm", "restart_fr_on_each_reply", 0)
> modparam("tm", "onreply_avp_mode", 1)
>
> #### Record Route Module
> loadmodule "rr.so"
> /* do not append from tag to the RR (no need for this script) */
> modparam("rr", "append_fromtag", 0)
>
> ### add for freeswitch
> loadmodule "dialog.so"
> modparam("dialog", "db_mode", 1)
> modparam("dialog", "db_url", 
> "postgres://opensips:opensipsrw at 192.168.113.145/opensips 
> <http://opensips:opensipsrw at 192.168.113.145/opensips>")
>
>
> #### MAX ForWarD module
> loadmodule "maxfwd.so"
>
> ### add for freeswitch
> loadmodule "textops.so"
>
>
> #### SIP MSG OPerationS module
> loadmodule "sipmsgops.so"
>
> #### FIFO Management Interface
> loadmodule "mi_fifo.so"
> modparam("mi_fifo", "fifo_name", "/run/opensips/opensips_fifo")
> # not working either, modparam("mi_fifo", "fifo_name", 
> "/tmp/opensips_fifo")
> modparam("mi_fifo", "fifo_mode", 0666)
>
> ### add for freeswitch
> loadmodule "dispatcher.so"
> modparam("dispatcher", "db_url", 
> "postgres://opensips:opensipsrw at 192.168.113.145/opensips 
> <http://opensips:opensipsrw at 192.168.113.145/opensips>")
> modparam("dispatcher", "ds_ping_method", "OPTIONS")
> modparam("dispatcher", "ds_ping_interval", 5)
> modparam("dispatcher", "ds_probing_threshhold", 2)
> modparam("dispatcher", "ds_probing_mode", 1)
>
> ### add for freeswitch
> loadmodule "load_balancer.so"
> modparam("load_balancer", "db_url", 
> "postgres://opensips:opensipsrw at 192.168.113.145/opensips 
> <http://opensips:opensipsrw at 192.168.113.145/opensips>")
> modparam("load_balancer", "probing_method", "OPTIONS")
> modparam("load_balancer", "probing_interval", 5)
>
>
>
> #### USeR LOCation module
> loadmodule "usrloc.so"
> modparam("usrloc", "nat_bflag", "NAT")
> modparam("usrloc", "working_mode_preset", "single-instance-no-db")
>
> #### REGISTRAR module
> loadmodule "registrar.so"
> modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
> /* uncomment the next line not to allow more than 10 contacts per AOR */
> #modparam("registrar", "max_contacts", 10)
>
> #### ACCounting module
> loadmodule "acc.so"
> /* what special events should be accounted ? */
> modparam("acc", "early_media", 0)
> modparam("acc", "report_cancels", 0)
> /* by default we do not adjust the direct of the sequential requests.
>    if you enable this parameter, be sure to enable "append_fromtag"
>    in "rr" module */
> modparam("acc", "detect_direction", 0)
>
> loadmodule "proto_udp.so"
>
>
> #### gjw add for rtpengine
> loadmodule "rtpengine.so"
> # single rtproxy
> modparam("rtpengine", "rtpengine_sock", "udp:192.168.113.143:2223 
> <http://192.168.113.143:2223>")
> # multiple rtproxies for LB
> #modparam("rtpengine", "rtpengine_sock", "udp:localhost:12221 
> udp:localhost:12222")
> # multiple sets of multiple rtproxies
> #modparam("rtpengine", "rtpengine_sock", "1 == udp:localhost:12221 
> udp:localhost:12222")
> #modparam("rtpengine", "rtpengine_sock", "2 == udp:localhost:12225")
>
> ### add for freeswitch
>
>
>
> ####### Routing Logic ########
>
> # main request routing logic
>
> 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");
> #                # Route to FreeSWITCH
> #                $du = "sip:192.168.113.142:5060 
> <http://192.168.113.142:5060>";
> #                forward();
>
> if (!load_balance(1,"pstn","1")) {
>     log("sreekanth invite method\n");
>     send_reply(503,"Service Unavailable");
>     exit;
> }
>
> }
>
>
> if (!is_myself("$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")) {
> # # store the registration and generate a SIP reply
> # if (!save("location"))
> # xlog("failed to register AoR $tu\n");
> #
> # exit;
>
>
> if (!ds_select_dst(1, 4)) {
>     send_reply(503,"Service Unavailable");
>     exit;
> }
> }
>
> # if ($rU==NULL) {
> # # request with no Username in RURI
> # send_reply(484,"Address Incomplete");
> # exit;
> # }
>
> # do lookup with method filtering
> if (!lookup("location","method-filtering")) {
> t_reply(404, "Not Found");
> exit;
> }
>
> # when routing via usrloc, log the missed calls also
> do_accounting("log","missed");
> route(relay);
> }
>
>
> route[relay] {
> # for INVITEs enable some additional helper routes
> if (is_method("INVITE")) {
> t_on_branch("per_branch_ops");
> t_on_reply("handle_nat");
> t_on_failure("missed_call");
> }
>
> if (!t_relay()) {
> send_reply(500,"Internal Error");
> }
> exit;
> }
>
>
>
>
> branch_route[per_branch_ops] {
> xlog("new branch at $ru\n");
> }
>
>
> onreply_route[handle_nat] {
> xlog("incoming reply\n");
> }
>
>
> failure_route[missed_call] {
> if (t_was_cancelled()) {
> exit;
> }
>
> # uncomment the following lines if you want to block client
> # redirect based on 3xx replies.
> ##if (t_check_status("3[0-9][0-9]")) {
> ##t_reply(404,"Not found");
> ## exit;
> ##}
>
>
> }
>
> ```
>
> Any idea why INVITE not working? :-)
>
>
> Nine to one <9to1url at gmail.com> 於 2023年9月14日 週四 下午4:40寫道：
>
>     Get idea from another thread: Reading documents :-)
>     Did that.
>
>     Nine to one <9to1url at gmail.com> 於 2023年9月14日 週四 上午9:12寫道：
>
>         Can anyone share the opensips conf file with Freeswitch and
>         rtpengine?
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From prathibhab.tvm at gmail.com  Fri Sep 15 05:37:27 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Fri, 15 Sep 2023 11:07:27 +0530
Subject: [OpenSIPS-Users] is_myself()
Message-ID: <CAJh_-cAE+=oQdVo+0QMTgAQyAYeMNKNfqtwhQT-tR=T5yy-UGg@mail.gmail.com>

I've changed
*from_uri!=myself to !is_myself($fu)*
*!uri==myself to !is_myself($ru)*
*!uri==myself to !is_myself($ru)*


*Is it correct?*

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Tue Sep 19 08:09:13 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 19 Sep 2023 13:39:13 +0530
Subject: [OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy replied
	with error: Unknown call-id
Message-ID: <CAJh_-cBEV7Bgx_2Ct6A_Qm1S9+pgSXhvsRv4JcEdUgwL6dTtvA@mail.gmail.com>

I am getting this error:

ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
call-id

How to resolve this?

I've attached the opensips.cf file with this email.

-- 
Regards,
B.Prathibha
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####### Global Parameters #########

#debug_mode=yes

log_level=3
xlog_level=3
log_stderror=no
log_facility=LOG_LOCAL0

udp_workers=4

#disable_dns_blacklist=no

#dns_try_ipv6=yes


socket=udp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tcp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tls:172.31.34.24:5061 as 65.2.167.22:5061  # CUSTOMIZE ME
socket=ws:172.31.34.24:8080 as 65.2.167.22:8080
socket=wss:172.31.34.24:7443 as 65.2.167.22:7443

####### Modules Section ########

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
loadmodule "proto_tls.so"
loadmodule "proto_ws.so"
loadmodule "proto_wss.so"
modparam("proto_wss", "wss_port", 7443)
modparam("proto_wss", "wss_max_msg_chunks", 16)

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### MYSQL module
loadmodule "db_mysql.so"

#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back")
modparam("usrloc", "db_url",
	"mysql://root:root at localhost/opensips") # CUSTOMIZE ME

#### avpops module
loadmodule "avpops.so"
modparam("avpops", "db_url",
	"mysql://root:root at localhost/opensips") # CUSTOMIZE ME


#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)

modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
	"mysql://root:root at localhost/opensips") # CUSTOMIZE ME

#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
	"mysql://root:root at localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")

loadmodule "alias_db.so"
modparam("alias_db", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
	"mysql://root:root at localhost/opensips") # CUSTOMIZE ME

####  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger at 127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")

####  DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

####  MI_HTTP module
loadmodule "mi_http.so"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so" 

loadmodule "tls_openssl.so"
#loadmodule "tls_wolfssl.so"
loadmodule "tls_mgm.so"
modparam("tls_mgm","server_domain", "default")
modparam("tls_mgm","verify_cert", "[default]0")
modparam("tls_mgm","require_cert", "[default]0")
#modparam("tls_mgm","certificate", "[default]/etc/opensips/tls/rootCA/cacert.pem")
#modparam("tls_mgm","private_key", "[default]/etc/opensips/tls/rootCA/private/cakey.pem")
#modparam("tls_mgm","ca_list", "[bp3.erss.in]/etc/opensips/tls/user/user-calist.pem")
modparam("tls_mgm", "tls_method", "[default]SSLv23")
modparam("tls_mgm","tls_library","openssl")
#modparam("tls_mgm", "certificate", "[default]/etc/opensips/tls/user/caKey.pem")
#modparam("tls_mgm", "private_key", "[default]/etc/opensips/tls/user/privateKey.pem")
#modparam("tls_mgm", "client_domain", "14.139.
modparam("tls_mgm", "certificate", "[default]/home/ubuntu/cert.pem")
modparam("tls_mgm", "private_key", "[default]/home/ubuntu/privkey.pem")

loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:172.31.34.24:2225")
####### Routing Logic ########

# main request routing logic

route{

	# initial NAT handling; detect if the request comes from behind a NAT
        # and apply contact fixing
        force_rport();
        if (nat_uac_test(23)) {
                if (is_method("REGISTER")) {
                        fix_nated_register();
                        setbflag("NAT");
                } else {
                        fix_nated_contact();
                        setflag("NAT");
                }
        }

	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;
		}

		# validate the sequential request against dialog
		if ( $DLG_status!=NULL && !validate_dialog() ) {
			xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
			## exit;
		}

		if (is_method("BYE")) {
			# do accounting even if the transaction fails
			do_accounting("db","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")) {
					
			# authenticate if from local subscriber
			# authenticate all initial non-REGISTER request that pretend to be
			# generated by local subscriber (domain from FROM URI is local)
			if (!proxy_authorize("", "subscriber")) {
				proxy_challenge("", "auth");
				exit;
			}
			if ($au!=$fU) {
				send_reply(403,"Forbidden auth ID");
				exit;
			}

			consume_credentials();
			# caller authenticated
			
		} 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");
		if (is_method("INVITE")) {
                        # even if in most of the cases is useless, do RR for
                        # re-INVITEs alos, as some buggy clients do change route set
                        # during the dialog.
                        record_route();
                }
		exit;
	}

	# record routing
	if (!is_method("REGISTER|MESSAGE"))
		record_route();

	# account only INVITEs
	if (is_method("INVITE")) {
		
		# create dialog with timeout
		if ( !create_dialog("B") ) {
			send_reply(500,"Internal Server Error");
			exit;
		}
		
		do_accounting("db");
		
	}

	
	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")) {
		xlog("Do REGISTER AUTH : [$rm] ci[$ci] si[$si] sp[$sp] rd[$rd] rU[$rU] fU[$fU]");
		# authenticate the REGISTER requests
                if (!www_authorize("", "subscriber")) {
                        www_challenge("", "auth");
                        exit;
                }

                if ($au!=$tU) {
                        send_reply(403,"Forbidden auth ID");
                        exit;
                }
                if ($socket_in(proto) == "tcp" || $socket_in(proto) == "tls")
                        setflag("TCP_PERSISTENT");
                if (isflagset("NAT")) {
                        setbflag("SIP_PING_FLAG");
                }
                # store the registration and generate a SIP reply
                if (!save("location")){
                        sl_reply_error();
                        xlog("failed to register AoR $tu\n");
                        exit;
                }

                exit;
		#route(AUTH);
		
	}

	if ($rU==NULL) {
		# request with no Username in RURI
		send_reply(484,"Address Incomplete");
		exit;
	}

	# apply DB based aliases
        alias_db_lookup("dbaliases");


        # apply transformations from dialplan table
        dp_translate( 0, "$rU", $rU);

        # check if the clients are using WebSockets or WebSocketSecure
        if ($socket_in(proto) == "WS" || $socket_in(proto) == "WSS")
                setflag('SRC_WS');
        else
                setflag('SRC_SIP');


        # consider the client is behind NAT - always fix the contact
        fix_nated_contact();

	if (is_method("REGISTER")) {

                # indicate that the client supports DTLS
                # so we know when he is called
                if (isflagset('SRC_WS'))
                        setbflag('DST_WS');

                fix_nated_register();
                if (!save("location"))
                        sl_reply_error();

                exit;
        }

	# do lookup with method filtering
	if (!lookup("location","m")) {
		if (!db_does_uri_exist("$ru","subscriber")) {
			send_reply(420,"Bad Extension");
			exit;
		}
		
		t_reply(404, "Not Found");
		exit;
	} 

	

	# when routing via usrloc, log the missed calls also
	do_accounting("db","missed");
	
	route(relay);
}

route[AUTH]{
	# authenticate the REGISTER requests
		$var(authRslt) = www_authorize("", "subscriber"); 
        xlog("register auth result [$var(authRslt)] rd [$rd] user[$fU] tu[$tu] source ip[$si]");
		switch ($var(authRslt)) {
    		  case -1:
        		send_reply(404, "Not Found");
        		exit;
    		  case -2:
    		  case -5:
        		send_reply(403, "Forbidden");
       			 exit;
    		  case -3:
    		  case -4:
        		www_challenge("","auth");  
        		exit;
		}
		if ($au!=$tU) {
			send_reply(403,"Forbidden auth ID");
			exit;
		}
		if ($socket_in(proto) == "tcp")
			setflag("TCP_PERSISTENT");
	

	 	avp_db_query("select count(*) from location where username = '$fU' and contact like 'sip:$fU@$si%'", "$avp(existExtenCount)");
		if ( $avp(existExtenCount) < 1 ) {
			xlog("SIP contact ct:[$ct] tu:[$tu] si[$si] did not registe on, then check registe status for tU:[$tU].");

			avp_db_query("select count(*) from location where username = '$fU'", "$avp(ct4fU)");
			if ( $avp(ct4fU) > 0) {
				xlog("L_WARN", "Forbid $ct to registe on, cuase by : exist another $fU has registed");
				send_reply(403, "Occupied");
				exit;
			}
		}else{
			xlog("--->SIP contact ct:[$ct] tu:[$tu] si[$si] is registe on, then update registe status for tU:[$tU].");
		}		
		# store the registration and generate a SIP reply
		if (!save("location"))
			xlog("failed to register AoR $tu\n");

		exit;
}


route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		
		

		#t_on_branch("per_branch_ops");
		t_on_branch("handle_nat");
		t_on_reply("handle_nat");
		t_on_failure("missed_call");
	}

	

	if (!t_relay()) {
		send_reply(500,"Internal Error");
	}
	exit;
}




#branch_route[per_branch_ops] {
#branch_route[handle_nat] {
#	  xlog("------------> SCRIPT:DBG BRANCH $T_branch_idx, $oP request $ru contact(s) $ct\n$cs\n $rm: message flags: $mf\n branch flags: $bf\n");
 # 	if(has_body("application/sdp")) {
  #      	$var(rtpengine_flags) = "RTP/AVP";
        	#$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
  #      	xlog("L_INFO", "INFO: RTPengine options: $var(rtpengine_flags)");
#	        rtpengine_offer("$var(rtpengine_flags)");
#       		rtpengine_play_media("file=/root/rtpengine/media/std_audio_file.wav");
#	}
	
#}


#onreply_route[handle_nat] {
#  fix_nated_contact();
#  xlog("L_DBG","SCRIPT:DBG $var(reply_log) $cs: $rs: response\nmessage flags: $mf\nbranch flags: $bf\n");
#  if(has_body("application/sdp")) {
#       $var(rtpengine_flags) = "RTP/AVP";
       # $var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
#        xlog("L_INFO", "INFO: RTPengine options: $var(rtpengine_flags)");
#        rtpengine_answer("$var(rtpengine_flags)");
#    }	
#}

branch_route[handle_nat] {

	if (!is_method("INVITE") || !has_body("application/sdp"))
		return;

	if (isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_offer("$var(rtpengine_flags)");
	rtpengine_start_recording();
}

onreply_route[handle_nat] {

	fix_nated_contact();
	if (!has_body("application/sdp"))
		return;

	if (isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_answer("$var(rtpengine_flags)");
	rtpengine_start_recording();
}

failure_route[missed_call] {
	if (t_was_cancelled()) {
		exit;
	}

	# uncomment the following lines if you want to block client 
	# redirect based on 3xx replies.
	##if (t_check_status("3[0-9][0-9]")) {
	##t_reply(404,"Not found");
	##	exit;
	##}

	
}



local_route {
	if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
		
		acc_db_request("200 Dialog Timeout", "acc");
		
	}
}

From bogdan at opensips.org  Tue Sep 19 08:20:23 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:20:23 +0300
Subject: [OpenSIPS-Users] Call center issue in 3.1;
 call center not working after migration to 3.2
In-Reply-To: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
References: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
Message-ID: <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>

Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function 
in the script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

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

On 7/4/23 1:15 PM, Kosmas Palios wrote:
>
> Hello community,
>
>
> Our team has managed to setup a SIP Trunk to forward calls to a number 
> of SIP clients, in opensips 3.1.16. We are using the call center 
> module, and it works fine for low traffic. We would like to ask two 
> separate but related questions.
>
>
> ISSUE 1
>
>
> We are using the call center module to forward calls to 100 SIP 
> agents, and it works well if traffic is relatively low (about 25 
> incoming calls per minute). However, when traffic is higher, i.e. up 
> to 60 incoming calls per minute, we see calls getting rejected because 
> of cc_handle_call() failing with error message:
>
>
> DBG:b2b_entities:server_new: It is a retransmission, drop
>
> ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
> server instance
>
>
> Unfortunately, every time this happens, an agent's status gets stuck 
> to "incall" forever, even though no cc_calls row includes him. So that 
> agent is lost.
>
>
> We are running in UDP mode, using 6 UDP workers. I’m attaching the 
> configuration file as opensips_3_1_16.cfg
>
> I can share the whole setup if needed.
>
>
> ISSUE 2
>
>
> We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
> leak. When we got to migrating the call center, we read this blogpost: 
> https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 
> <https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/>
>
>
> " When comes to the modules using the b2b_logicAPI (providing features 
> on top of the B2B engine), the only affected one is the 
> call_centermodule. The change is minor – the xml file controlling the 
> call queuing logic was removed, as not needed any more. Otherwise, in 
> terms of usage, it is exactly the same."
>
>
> However, when we removed the lines:
>
>
> modparam("b2b_logic_xml","script_scenario", 
> "/etc/opensips/scenario_callcenter.xml")
>
> modparam("call_center", "b2b_scenario", "call center")
>
>
> the call center started behaving weird: it created another invite to 
> the sip trunk, instead of creating the invite to the agent (the call 
> id was good, but the to uri was wrong). I can give detailed logs on 
> this, but I wouldn't want to make this email any bigger than it 
> already is. I’m also attaching the configuration file as 
> opensips_3_2_13.cfg
>
>
> To sum up, our questions are:
>
>
> 1. Any ideas on what the problem is with creating a new server 
> instance for high numbers of calls?
>
> 2. What's the recommended way to migrate the call center to version 
> 3.2 ? Can we find an example script-driven call center somewhere?
>
>
> Thank you in advance for your help!
>
> Best regards,
>
> Kosmas
>
>
> P.S.: about our team: we are a small team from Athens, Greece 
> integrating voice assistants on various platforms. Unfortunately we 
> missed the latest Opensips summit held last September.
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Tue Sep 19 08:28:46 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:28:46 +0300
Subject: [OpenSIPS-Users] Implementation Issues with Opensips as a Load
 Balancer
In-Reply-To: <CAH58Vz9bu3A4B6XqLGcbwS5TPLjbk709sYneFmpLrf4oxHQkmA@mail.gmail.com>
References: <CAH58Vz9bu3A4B6XqLGcbwS5TPLjbk709sYneFmpLrf4oxHQkmA@mail.gmail.com>
Message-ID: <baef9cfe-ad69-1c4b-b596-ffffbea5a913@opensips.org>

Hi Joan,

A typically LB does not handle registrations, is just doing pass thru 
for the calls. Again the LB is just balancing the calls between multiple 
back servers, it does not handle REGISTER traffic

Best regards,

Bogdan-Andrei Iancu

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

On 7/10/23 9:22 PM, Joan Leyrana wrote:
>
> Hello Team, I'm trying to implement Opensips with Asterisk as a load 
> balancer (I have been looking at the documentation, but I can't find a 
> guide for the newer versions of Opensips). I have already installed 
> the MySQL database, Opensips version 3.3, and OCP 9.3.3. I have added 
> my Asterisk servers to the load balancer, generated my load balancer 
> script, and activated the necessary modules in the config file. 
> However, it is not working for me, and I have the following doubts:
>
> Will the VoIP phones register with Opensips? Why does generating the 
> load balancer script disable the registrar module?
>
> The route logic generated by the load balancer is not working, or at 
> least it doesn't allow me to register the phones.
>
> My understanding is that the route is as follows:
>
> VoIP Phone -- >Registrar --> Opensips Opensips --> Media --> Asterisk
>
> Am I skipping any important steps
>
> Regards,
>
>
> Banner <https://www.nextor.io/>
> *Joan Leyrana*
> Gerente Infraestructura | Nextor Telecom
> ** *O *55 4440 6008
> *E *joan at nextor.io <mailto:joan at nextor.io>
> Presa Falcón 128 Irrigación, CDMX 11500
> *www.nextor.io* <http://www.nextor.io/>
>
> 	
> 	
> facebook icon <https://www.facebook.com/nextortelecom>
> twitter icon <https://twitter.com/Nextor_oficial>
> youtube icon <https://www.youtube.com/channel/UCjnr6hlUyIntPspNizXOeIg>
>
> 	
> linkedin icon <https://www.linkedin.com/company/nextor-telecom/>
> instagram icon <https://www.instagram.com/nextortelecom/>
>
> 	
> Banner <https://yotellamo.lanubesota.mx/2004670>
> Banner 
> <https://api.whatsapp.com/send/?phone=525514540020&text=Hola%21Nextor!%2C+tu+firma+de+correo+me+manda+a+tu+chat&app_absent=0> 
>
> Banner <https://nextor.io/img-firmas-email/Nextor_Logo.png>
>
> Contacto con supervisor : *calidad at nextor.io <mailto:calidad at nextor.io>*
>
> Considere el medio ambiente antes de imprimir este mensaje. 
> *CONFIDENCIALIDAD.* Este mensaje puede contener información 
> confidencial, por lo que su divulgación está prohibida de acuerdo a 
> las leyes aplicables. Si usted no es el destinatario o el responsable 
> de la entrega del presente, se le notifica que la publicación, 
> distribución o reproducción de este mensaje, quedan estrictamente 
> prohibidas. *AVISO DE PRIVACIDAD*. NZXT TELECOMUNICACIONES DE MEXICO, 
> S.A. DE C.V., con domicilio en Presa Falcón 128 Primer piso, Colonia 
> Irrigación, CDMX 11500, utilizará sus datos personales recabados para 
> proveer y/o realizar gestiones administrativas relacionadas con los 
> servicios y/o productos que ofrece. Para mayor información acerca del 
> tratamiento de los derechos que puede hacer valer, puede acceder al 
> Aviso de Privacidad Integral a través de la página de internet 
> www.nextor.io <http://www.nextor.io>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Tue Sep 19 08:32:23 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:32:23 +0300
Subject: [OpenSIPS-Users] is_myself()
In-Reply-To: <CAJh_-cAE+=oQdVo+0QMTgAQyAYeMNKNfqtwhQT-tR=T5yy-UGg@mail.gmail.com>
References: <CAJh_-cAE+=oQdVo+0QMTgAQyAYeMNKNfqtwhQT-tR=T5yy-UGg@mail.gmail.com>
Message-ID: <1321a75a-a77b-bfbb-61ca-dc0eba374c8b@opensips.org>

The documentation is your best friend,:
https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#is_myself

Best regards,

Bogdan-Andrei Iancu

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

On 9/15/23 8:37 AM, Prathibha B wrote:
> I've changed
> *from_uri!=myself to !is_myself($fu)*
> *!uri==myself to !is_myself($ru)*
> *!uri==myself to !is_myself($ru)*
>
> *Is it correct?
> *
>
> -- 
> Regards,
> B.Prathibha
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From kosmas.palios at athenarc.gr  Tue Sep 19 08:35:22 2023
From: kosmas.palios at athenarc.gr (Kosmas Palios)
Date: Tue, 19 Sep 2023 08:35:22 +0000
Subject: [OpenSIPS-Users] Call center issue in 3.1;
 call center not working after migration to 3.2
In-Reply-To: <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>
References: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
 <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>
Message-ID: <DBBPR05MB65063DDCF8BE3D6303EA8E4DE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>

This is indeed the solution, as we figured out by ourselves and as documented in the tm documentation!

Thank you for your kind reply here, as well as at the issue I posted sometime ago i github (my username is kosniaz).

Any input to the second issue (raised on github as well about 10 days ago) would be greatly appreciated!

Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Users <users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:20:23 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas Palios <kosmas.palios at gmail.com>
Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>
Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function in the script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

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

On 7/4/23 1:15 PM, Kosmas Palios wrote:

Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number of SIP clients, in opensips 3.1.16. We are using the call center module, and it works fine for low traffic. We would like to ask two separate but related questions.


ISSUE 1


We are using the call center module to forward calls to 100 SIP agents, and it works well if traffic is relatively low (about 25 incoming calls per minute). However, when traffic is higher, i.e. up to 60 incoming calls per minute, we see calls getting rejected because of cc_handle_call() failing with error message:


DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server instance


Unfortunately, every time this happens, an agent's status gets stuck to "incall" forever, even though no cc_calls row includes him. So that agent is lost.


We are running in UDP mode, using 6 UDP workers. I’m attaching the configuration file as opensips_3_1_16.cfg

I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients leak. When we got to migrating the call center, we read this blogpost: https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/


" When comes to the modules using the b2b_logic API (providing features on top of the B2B engine), the only affected one is the call_center module. The change is minor – the xml file controlling the call queuing logic was removed, as not needed any more. Otherwise, in terms of usage, it is exactly the same."


However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", "/etc/opensips/scenario_callcenter.xml")

modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to the sip trunk, instead of creating the invite to the agent (the call id was good, but the to uri was wrong). I can give detailed logs on this, but I wouldn't want to make this email any bigger than it already is. I’m also attaching the configuration file as opensips_3_2_13.cfg


To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server instance for high numbers of calls?

2. What's the recommended way to migrate the call center to version 3.2 ? Can we find an example script-driven call center somewhere?

Thank you in advance for your help!

Best regards,

Kosmas



P.S.: about our team: we are a small team from Athens, Greece integrating voice assistants on various platforms. Unfortunately we missed the latest Opensips summit held last September.




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


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From bogdan at opensips.org  Tue Sep 19 08:42:03 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:42:03 +0300
Subject: [OpenSIPS-Users] Call center issue in 3.1;
 call center not working after migration to 3.2
In-Reply-To: <DBBPR05MB65063DDCF8BE3D6303EA8E4DE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
References: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
 <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>
 <DBBPR05MB65063DDCF8BE3D6303EA8E4DE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
Message-ID: <924f3395-b4f8-4afc-ae0f-43e4f5c14b14@opensips.org>

For the second issue, could you point the GH ticket?

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:35 AM, Kosmas Palios wrote:
> This is indeed the solution, as we figured out by ourselves and as 
> documented in the tm documentation!
>
> Thank you for your kind reply here, as well as at the issue I posted 
> sometime ago i github (my username is kosniaz).
>
> Any input to the second issue (raised on github as well about 10 days 
> ago) would be greatly appreciated!
>
> Thank you very much one more time for your insights.
>
> Have a great day!
>
> Kosmas
>
> Sent from Outlook for iOS <https://aka.ms/o0ukef>
> ------------------------------------------------------------------------
> *From:* Users <users-bounces at lists.opensips.org> on behalf of 
> Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Sent:* Tuesday, September 19, 2023 11:20:23 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas 
> Palios <kosmas.palios at gmail.com>
> *Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
> not working after migration to 3.2
>
> 	
> Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
> bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό 
> <https://aka.ms/LearnAboutSenderIdentification>
> 	
>
> Hi Kosmas,
>
> For Issue 1, try placing a t_newtran(); before calling the cc function 
> in the script - this will prevent the issues due to retransmissions.
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>    https://www.siphub.com  <https://www.siphub.com>
> On 7/4/23 1:15 PM, Kosmas Palios wrote:
>>
>> Hello community,
>>
>>
>> Our team has managed to setup a SIP Trunk to forward calls to a 
>> number of SIP clients, in opensips 3.1.16. We are using the call 
>> center module, and it works fine for low traffic. We would like to 
>> ask two separate but related questions.
>>
>>
>> ISSUE 1
>>
>>
>> We are using the call center module to forward calls to 100 SIP 
>> agents, and it works well if traffic is relatively low (about 25 
>> incoming calls per minute). However, when traffic is higher, i.e. up 
>> to 60 incoming calls per minute, we see calls getting rejected 
>> because of cc_handle_call() failing with error message:
>>
>>
>> DBG:b2b_entities:server_new: It is a retransmission, drop
>>
>> ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
>> server instance
>>
>>
>> Unfortunately, every time this happens, an agent's status gets stuck 
>> to "incall" forever, even though no cc_calls row includes him. So 
>> that agent is lost.
>>
>>
>> We are running in UDP mode, using 6 UDP workers. I’m attaching the 
>> configuration file as opensips_3_1_16.cfg
>>
>> I can share the whole setup if needed.
>>
>>
>> ISSUE 2
>>
>>
>> We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
>> leak. When we got to migrating the call center, we read this 
>> blogpost: 
>> https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 
>> <https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/>
>>
>>
>> " When comes to the modules using the b2b_logicAPI (providing 
>> features on top of the B2B engine), the only affected one is the 
>> call_centermodule. The change is minor – the xml file controlling the 
>> call queuing logic was removed, as not needed any more. Otherwise, in 
>> terms of usage, it is exactly the same."
>>
>>
>> However, when we removed the lines:
>>
>>
>> modparam("b2b_logic_xml","script_scenario", 
>> "/etc/opensips/scenario_callcenter.xml")
>>
>> modparam("call_center", "b2b_scenario", "call center")
>>
>>
>> the call center started behaving weird: it created another invite to 
>> the sip trunk, instead of creating the invite to the agent (the call 
>> id was good, but the to uri was wrong). I can give detailed logs on 
>> this, but I wouldn't want to make this email any bigger than it 
>> already is. I’m also attaching the configuration file as 
>> opensips_3_2_13.cfg
>>
>>
>> To sum up, our questions are:
>>
>>
>> 1. Any ideas on what the problem is with creating a new server 
>> instance for high numbers of calls?
>>
>> 2. What's the recommended way to migrate the call center to version 
>> 3.2 ? Can we find an example script-driven call center somewhere?
>>
>>
>> Thank you in advance for your help!
>>
>> Best regards,
>>
>> Kosmas
>>
>>
>> P.S.: about our team: we are a small team from Athens, Greece 
>> integrating voice assistants on various platforms. Unfortunately we 
>> missed the latest Opensips summit held last September.
>>
>>
>>
>> _______________________________________________
>> 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

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From bogdan at opensips.org  Tue Sep 19 08:43:10 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:43:10 +0300
Subject: [OpenSIPS-Users] Increased macro MAX_BRANCHES and behavior of
 tm module
In-Reply-To: <CAJUEF2uhxMBW-vLLarr1Demo7ypHnja5YQt3BoWhBAZ9Wd=ZDQ@mail.gmail.com>
References: <CAJUEF2uhxMBW-vLLarr1Demo7ypHnja5YQt3BoWhBAZ9Wd=ZDQ@mail.gmail.com>
Message-ID: <b5d4cf13-433a-6573-493f-02f90b49e15c@opensips.org>

Hi,

The 31 is a hard limit as the TM module is internally using an integer 
for storing a branch bitmask. Here is the limitation coming from. The 
change is not a trivial one, still not impossible (to drop the int 
bitmask and move to an array of int)

Regards,

Bogdan-Andrei Iancu

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

On 9/7/23 2:49 PM, Dmitry Ponomaryov wrote:
>
> Have a nice day, everyone
>
> Question is to increase the value of MAX_BRANCHES[1] to 32, for 
> example, which will go beyond the boundaries of the source code in 
> tm.c, namely condition[2], which clearly shows that only 30 is 
> possible, how critical it is to change the condition in tm.c, in order 
> to get the need value? [1] 
> https://github.com/OpenSIPS/opensips/blob/master/config.h#L169 
> <https://github.com/OpenSIPS/opensips/blob/master/config.h#L169> [2] 
> https://github.com/OpenSIPS/opensips/blob/master/modules/tm/tm.c#L817-L821 
> <https://github.com/OpenSIPS/opensips/blob/master/modules/tm/tm.c#L817-L821>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Tue Sep 19 08:46:01 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:46:01 +0300
Subject: [OpenSIPS-Users] opensips deployment as load balancer for T.38
 traffic
In-Reply-To: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>
References: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>
Message-ID: <d758442b-e058-b763-bf95-531d182abd3c@opensips.org>

Hi,

Take a look here, it might be a good starting point:
     https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

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

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
> Interested to know how we can deploy opensips as load balancer for 
> T.38 calls , whether it will work fine or we need to use rabbitmq for 
> load balancing
>
> scenario is
>
> ICTFax => opesips load balancer => freeswitch nodes
>
>
>
> regards
> *Tahir Almas*
>
> Managing Partner
> ICT Innovations
> http://www.ictinnovations.com <http://www.ictinnovations.com>
> Leveraging open source in ICT
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From kosmas.palios at athenarc.gr  Tue Sep 19 08:47:24 2023
From: kosmas.palios at athenarc.gr (Kosmas Palios)
Date: Tue, 19 Sep 2023 08:47:24 +0000
Subject: [OpenSIPS-Users] Call center issue in 3.1;
 call center not working after migration to 3.2
In-Reply-To: <924f3395-b4f8-4afc-ae0f-43e4f5c14b14@opensips.org>
References: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
 <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>
 <DBBPR05MB65063DDCF8BE3D6303EA8E4DE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
 <924f3395-b4f8-4afc-ae0f-43e4f5c14b14@opensips.org>
Message-ID: <DBBPR05MB650643469FE9CB409E945761E9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>

of course, here it is

https://github.com/OpenSIPS/opensips/issues/3176

thank you



Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:42:03 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas Palios <kosmas.palios at athenarc.gr>; Kosmas Palios <kosmas.palios at gmail.com>
Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>
For the second issue, could you point the GH ticket?

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:35 AM, Kosmas Palios wrote:
This is indeed the solution, as we figured out by ourselves and as documented in the tm documentation!

Thank you for your kind reply here, as well as at the issue I posted sometime ago i github (my username is kosniaz).

Any input to the second issue (raised on github as well about 10 days ago) would be greatly appreciated!

Thank you very much one more time for your insights.

Have a great day!

Kosmas

Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Users <users-bounces at lists.opensips.org><mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:20:23 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Kosmas Palios <kosmas.palios at gmail.com><mailto:kosmas.palios at gmail.com>
Subject: Re: [OpenSIPS-Users] Call center issue in 3.1; call center not working after migration to 3.2


Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org<mailto:bogdan at opensips.org>. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>

Hi Kosmas,

For Issue 1, try placing a t_newtran(); before calling the cc function in the script - this will prevent the issues due to retransmissions.

Regards,

Bogdan-Andrei Iancu

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

On 7/4/23 1:15 PM, Kosmas Palios wrote:

Hello community,


Our team has managed to setup a SIP Trunk to forward calls to a number of SIP clients, in opensips 3.1.16. We are using the call center module, and it works fine for low traffic. We would like to ask two separate but related questions.


ISSUE 1


We are using the call center module to forward calls to 100 SIP agents, and it works well if traffic is relatively low (about 25 incoming calls per minute). However, when traffic is higher, i.e. up to 60 incoming calls per minute, we see calls getting rejected because of cc_handle_call() failing with error message:


DBG:b2b_entities:server_new: It is a retransmission, drop

ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b server instance


Unfortunately, every time this happens, an agent's status gets stuck to "incall" forever, even though no cc_calls row includes him. So that agent is lost.


We are running in UDP mode, using 6 UDP workers. I’m attaching the configuration file as opensips_3_1_16.cfg

I can share the whole setup if needed.


ISSUE 2


We decided to migrate to 3.2 after seeing the bugfix to b2b_clients leak. When we got to migrating the call center, we read this blogpost: https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/


" When comes to the modules using the b2b_logic API (providing features on top of the B2B engine), the only affected one is the call_center module. The change is minor – the xml file controlling the call queuing logic was removed, as not needed any more. Otherwise, in terms of usage, it is exactly the same."


However, when we removed the lines:


modparam("b2b_logic_xml","script_scenario", "/etc/opensips/scenario_callcenter.xml")

modparam("call_center", "b2b_scenario", "call center")


the call center started behaving weird: it created another invite to the sip trunk, instead of creating the invite to the agent (the call id was good, but the to uri was wrong). I can give detailed logs on this, but I wouldn't want to make this email any bigger than it already is. I’m also attaching the configuration file as opensips_3_2_13.cfg


To sum up, our questions are:


1. Any ideas on what the problem is with creating a new server instance for high numbers of calls?

2. What's the recommended way to migrate the call center to version 3.2 ? Can we find an example script-driven call center somewhere?

Thank you in advance for your help!

Best regards,

Kosmas



P.S.: about our team: we are a small team from Athens, Greece integrating voice assistants on various platforms. Unfortunately we missed the latest Opensips summit held last September.




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





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


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From bogdan at opensips.org  Tue Sep 19 08:57:16 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 11:57:16 +0300
Subject: [OpenSIPS-Users] rest_post logs
In-Reply-To: <CANoA6_+pO=UJwvMcbqop-_-eUBO5UkyyNXHp8LdNNuq8_j8TTA@mail.gmail.com>
References: <CANoA6_+pO=UJwvMcbqop-_-eUBO5UkyyNXHp8LdNNuq8_j8TTA@mail.gmail.com>
Message-ID: <8c4e361b-a28f-4bb8-2601-7c7c33ee2d46@opensips.org>

HI Alberto,

What OpenSIPS version do you have?

And are those the only log lines you get ?

Regards,

Bogdan-Andrei Iancu

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

On 9/1/23 4:43 PM, Alberto wrote:
> Hi,
>
> Is there a way to turn off logs for rest_post requests?
>
> This is my log settings:
> log_level=-1
> xlog_level=-1
> log_stdout=yes
> log_stderror=yes
> log_facility=LOG_LOCAL0
>
> This is the type of output I want to suppress:
> Sep 01 13:31:48 opensips opensips[59740]: > POST /api/opensips/doit 
> HTTP/1.1
> Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
> Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
> Sep 01 13:31:48 opensips opensips[59740]: Content-Type: application/json
> Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
> Sep 01 13:31:48 opensips opensips[59740]:
> Sep 01 13:31:48 opensips opensips[59740]: * upload completely sent 
> off: 156 out of 156 bytes
> Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not 
> supporting multiuse
> Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
> Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
> Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep 2023 
> 13:31:48 GMT
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Type: 
> application/json; charset=utf-8
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
> Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
> Sep 01 13:31:48 opensips opensips[59740]: < Access-Control-Allow-Origin: *
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Security-Policy: 
> default-src 'self';base-uri 'self';font-src 'self' https: 
> data:;form-action 'self';frame-ancestors 'self';img-src 'self' 
> data:;object-src 'none';script-src 'self';script-src-attr 
> 'none';style-src 'self' https: 'unsafe-inline';upgrade-insecure-requests
> Sep 01 13:31:48 opensips opensips[59740]: < 
> Cross-Origin-Opener-Policy: same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < 
> Cross-Origin-Resource-Policy: same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
> Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy: no-referrer
> Sep 01 13:31:48 opensips opensips[59740]: < Strict-Transport-Security: 
> max-age=15552000; includeSubDomains
> Sep 01 13:31:48 opensips opensips[59740]: < X-Content-Type-Options: 
> nosniff
> Sep 01 13:31:48 opensips opensips[59740]: < X-DNS-Prefetch-Control: off
> Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options: noopen
> Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options: SAMEORIGIN
> Sep 01 13:31:48 opensips opensips[59740]: < 
> X-Permitted-Cross-Domain-Policies: none
> Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
> Sep 01 13:31:48 opensips opensips[59740]: <
> Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host 
> 127.0.0.1 left intact
>
> Thanks
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From kosmas.palios at athenarc.gr  Tue Sep 19 08:58:36 2023
From: kosmas.palios at athenarc.gr (Kosmas Palios)
Date: Tue, 19 Sep 2023 08:58:36 +0000
Subject: [OpenSIPS-Users] opensips deployment as load balancer for T.38
 traffic
In-Reply-To: <d758442b-e058-b763-bf95-531d182abd3c@opensips.org>
References: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>
 <d758442b-e058-b763-bf95-531d182abd3c@opensips.org>
Message-ID: <DBBPR05MB65065A5E2B70407B0F78F32CE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>

Thank you for your response.

We have tried generating different configs. We suspect that the problem is in the call center scenario (possibly the absence of). There seems to be a problem in connecting the incoming call to a new client instance.

The b2b mod generated INVITE that has the right agent in the INVITE Header but the wrong <To> Header. We have looked into the debug messages and there seem to be no reason for this issue. I will post some useful debug lines here later.

Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Users <users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:46:01 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Tahir Almas Dhesi <tahir at ictinnovations.com>
Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>
Hi,

Take a look here, it might be a good starting point:
    https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

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

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for  T.38 calls , whether it will work fine or we need to use rabbitmq for load balancing

scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




_______________________________________________
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 prathibhab.tvm at gmail.com  Tue Sep 19 09:08:27 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 19 Sep 2023 14:38:27 +0530
Subject: [OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy
 replied with error: Unknown call-id
In-Reply-To: <CAJh_-cBEV7Bgx_2Ct6A_Qm1S9+pgSXhvsRv4JcEdUgwL6dTtvA@mail.gmail.com>
References: <CAJh_-cBEV7Bgx_2Ct6A_Qm1S9+pgSXhvsRv4JcEdUgwL6dTtvA@mail.gmail.com>
Message-ID: <CAJh_-cAZodsOAToQEcA+mPowsSgK09tgYZ2KqpidugAe3x3nBQ@mail.gmail.com>

Can anyone pls help?

On Tue, 19 Sept 2023 at 13:39, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I am getting this error:
>
> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
> call-id
>
> How to resolve this?
>
> I've attached the opensips.cf file with this email.
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From bogdan at opensips.org  Tue Sep 19 09:11:27 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 12:11:27 +0300
Subject: [OpenSIPS-Users] upgrade failed
In-Reply-To: <edbd3677-c1d2-e0a7-49b7-7595af38162d@bohboh.info>
References: <CAC9cSOBTr2z9kY5GJ=eqG1PeaNzBH2fO32A4JMNF7DQUniUORw@mail.gmail.com>
 <edbd3677-c1d2-e0a7-49b7-7595af38162d@bohboh.info>
Message-ID: <860efc08-ebe4-27dd-8fee-f09a2e3dcb4a@opensips.org>

Hi,

That;s not the recommended way to fix the issue, as the the structure of 
the `dispatcher ` table is actually different in 3.4 and 3.1. Bypassing 
the check may result in failing DB queries.

The proper fix is install the correct dispatcher table for 3.4:
https://github.com/OpenSIPS/opensips/blob/3.4/scripts/mysql/dispatcher-create.sql

Or use the db migration/create functionality of opensips-cli

Regards,

Bogdan-Andrei Iancu

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

On 9/10/23 10:02 AM, Social Boh wrote:
>
> mysql -u root -p
>
> use opensips
>
> update version set table_version='9' where table_name='dispatcher';
>
> quit
>
> ---
> I'm SoCIaL, MayBe
> El 10/09/2023 a las 12:33 a. m., Saint Michael escribió:
>> I attempted an upgrade to version 3.4 from 3.1 and It failed
>> ERROR:core:db_check_table_version: invalid version 8 for table 
>> dispatcher found, expected 9
>> ERROR:dispatcher:mod_init: failed to init database support
>> ERROR:core:init_mod: failed to initialize module dispatcher
>> ERROR:core:main: error while initializing modules
>>
>> how do I upgrade the dispatcher table from 8 version 8 to version 9?
>> The script is fine.
>>
>>
>>
>>
>> _______________________________________________
>> 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 bogdan at opensips.org  Tue Sep 19 09:13:51 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 12:13:51 +0300
Subject: [OpenSIPS-Users] opensips deployment as load balancer for T.38
 traffic
In-Reply-To: <DBBPR05MB65065A5E2B70407B0F78F32CE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
References: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>
 <d758442b-e058-b763-bf95-531d182abd3c@opensips.org>
 <DBBPR05MB65065A5E2B70407B0F78F32CE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
Message-ID: <3c4a02cd-36be-5800-fe25-2b47ecd5b1a3@opensips.org>

Not sure what the callcenter has to do here with the LB scenario. LB is 
simply distributing (according to the load) a call to a bunch of similar 
back servers.

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:58 AM, Kosmas Palios wrote:
> Thank you for your response.
>
> We have tried generating different configs. We suspect that the 
> problem is in the call center scenario (possibly the absence of). 
> There seems to be a problem in connecting the incoming call to a new 
> client instance.
>
> The b2b mod generated INVITE that has the right agent in the INVITE 
> Header but the wrong <To> Header. We have looked into the debug 
> messages and there seem to be no reason for this issue. I will post 
> some useful debug lines here later.
>
> Sent from Outlook for iOS <https://aka.ms/o0ukef>
> ------------------------------------------------------------------------
> *From:* Users <users-bounces at lists.opensips.org> on behalf of 
> Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Sent:* Tuesday, September 19, 2023 11:46:01 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; Tahir 
> Almas Dhesi <tahir at ictinnovations.com>
> *Subject:* Re: [OpenSIPS-Users] opensips deployment as load balancer 
> for T.38 traffic
>
> 	
> Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
> bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό 
> <https://aka.ms/LearnAboutSenderIdentification>
> 	
>
> Hi,
>
> Take a look here, it might be a good starting point:
> https://www.opensips.org/Documentation/Generating-Configs-3-2 
> <https://www.opensips.org/Documentation/Generating-Configs-3-2>
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>    https://www.siphub.com  <https://www.siphub.com>
> On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
>> Interested to know how we can deploy opensips as load balancer for 
>> T.38 calls , whether it will work fine or we need to use rabbitmq for 
>> load balancing
>>
>> scenario is
>>
>> ICTFax => opesips load balancer => freeswitch nodes
>>
>>
>>
>> regards
>> *Tahir Almas*
>>
>> Managing Partner
>> ICT Innovations
>> http://www.ictinnovations.com <http://www.ictinnovations.com>
>> Leveraging open source in ICT
>>
>>
>> _______________________________________________
>> 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

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From alberto.rinaudo at gmail.com  Tue Sep 19 09:14:35 2023
From: alberto.rinaudo at gmail.com (Alberto)
Date: Tue, 19 Sep 2023 10:14:35 +0100
Subject: [OpenSIPS-Users] rest_post logs
In-Reply-To: <8c4e361b-a28f-4bb8-2601-7c7c33ee2d46@opensips.org>
References: <CANoA6_+pO=UJwvMcbqop-_-eUBO5UkyyNXHp8LdNNuq8_j8TTA@mail.gmail.com>
 <8c4e361b-a28f-4bb8-2601-7c7c33ee2d46@opensips.org>
Message-ID: <CANoA6_K8JYTjKHr_PuYh6_Pd8kdkeX90-RPmu_p7Mf0kTwmaPw@mail.gmail.com>

Hi,

Thanks for your reply,

I'm using v3.2.
I realized that that's the output from libcurl, which I can turn off by
setting log_stdout and log_stderror to no.

Thanks again

p.s. Have you ever thought about a forum instead of the mailing list? I
would have gone back and answered/closed my own question, but I didn't know
if I should have done that via mail.


On Tue, 19 Sept 2023 at 09:57, Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> HI Alberto,
>
> What OpenSIPS version do you have?
>
> And are those the only log lines you get ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 9/1/23 4:43 PM, Alberto wrote:
>
> Hi,
>
> Is there a way to turn off logs for rest_post requests?
>
> This is my log settings:
> log_level=-1
> xlog_level=-1
> log_stdout=yes
> log_stderror=yes
> log_facility=LOG_LOCAL0
>
> This is the type of output I want to suppress:
> Sep 01 13:31:48 opensips opensips[59740]: > POST /api/opensips/doit
> HTTP/1.1
> Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
> Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
> Sep 01 13:31:48 opensips opensips[59740]: Content-Type: application/json
> Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
> Sep 01 13:31:48 opensips opensips[59740]:
> Sep 01 13:31:48 opensips opensips[59740]: * upload completely sent off:
> 156 out of 156 bytes
> Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not supporting
> multiuse
> Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
> Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
> Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep 2023
> 13:31:48 GMT
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Type:
> application/json; charset=utf-8
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
> Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
> Sep 01 13:31:48 opensips opensips[59740]: < Access-Control-Allow-Origin: *
> Sep 01 13:31:48 opensips opensips[59740]: < Content-Security-Policy:
> default-src 'self';base-uri 'self';font-src 'self' https:
> data:;form-action 'self';frame-ancestors 'self';img-src 'self'
> data:;object-src 'none';script-src 'self';script-src-attr
> 'none';style-src 'self' https: 'unsafe-inline';upgrade-insecure-requests
> Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Opener-Policy:
> same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < Cross-Origin-Resource-Policy:
> same-origin
> Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
> Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy: no-referrer
> Sep 01 13:31:48 opensips opensips[59740]: < Strict-Transport-Security:
> max-age=15552000; includeSubDomains
> Sep 01 13:31:48 opensips opensips[59740]: < X-Content-Type-Options: nosniff
> Sep 01 13:31:48 opensips opensips[59740]: < X-DNS-Prefetch-Control: off
> Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options: noopen
> Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options: SAMEORIGIN
> Sep 01 13:31:48 opensips opensips[59740]: <
> X-Permitted-Cross-Domain-Policies: none
> Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
> Sep 01 13:31:48 opensips opensips[59740]: <
> Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host
> 127.0.0.1 left intact
>
> Thanks
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From bogdan at opensips.org  Tue Sep 19 09:20:10 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 12:20:10 +0300
Subject: [OpenSIPS-Users] Call center issue in 3.1;
 call center not working after migration to 3.2
In-Reply-To: <DBBPR05MB650643469FE9CB409E945761E9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
References: <CA+iUSH3GRTyhTyW=wrwmFvyA9boH3CZ60a4pY_RdtSBPNRq3cw@mail.gmail.com>
 <2561a201-b440-3153-45cf-e97901854ca9@opensips.org>
 <DBBPR05MB65063DDCF8BE3D6303EA8E4DE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
 <924f3395-b4f8-4afc-ae0f-43e4f5c14b14@opensips.org>
 <DBBPR05MB650643469FE9CB409E945761E9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
Message-ID: <b7c9a8a2-dbdf-19d5-1b69-ffb2d32af22a@opensips.org>

Thanks, let's move the discussion over there.

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:47 AM, Kosmas Palios wrote:
> of course, here it is
>
> https://github.com/OpenSIPS/opensips/issues/3176 
> <https://github.com/OpenSIPS/opensips/issues/3176>
>
> thank you
>
>
>
> Sent from Outlook for iOS <https://aka.ms/o0ukef>
> ------------------------------------------------------------------------
> *From:* Bogdan-Andrei Iancu <bogdan at opensips.org>
> *Sent:* Tuesday, September 19, 2023 11:42:03 AM
> *To:* OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas 
> Palios <kosmas.palios at athenarc.gr>; Kosmas Palios 
> <kosmas.palios at gmail.com>
> *Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
> not working after migration to 3.2
>
> 	
> Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
> bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό 
> <https://aka.ms/LearnAboutSenderIdentification>
> 	
>
> For the second issue, could you point the GH ticket?
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>    https://www.siphub.com  <https://www.siphub.com>
> On 9/19/23 11:35 AM, Kosmas Palios wrote:
>> This is indeed the solution, as we figured out by ourselves and as 
>> documented in the tm documentation!
>>
>> Thank you for your kind reply here, as well as at the issue I posted 
>> sometime ago i github (my username is kosniaz).
>>
>> Any input to the second issue (raised on github as well about 10 days 
>> ago) would be greatly appreciated!
>>
>> Thank you very much one more time for your insights.
>>
>> Have a great day!
>>
>> Kosmas
>>
>> Sent from Outlook for iOS <https://aka.ms/o0ukef>
>> ------------------------------------------------------------------------
>> *From:* Users <users-bounces at lists.opensips.org> 
>> <mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei 
>> Iancu <bogdan at opensips.org> <mailto:bogdan at opensips.org>
>> *Sent:* Tuesday, September 19, 2023 11:20:23 AM
>> *To:* OpenSIPS users mailling list <users at lists.opensips.org> 
>> <mailto:users at lists.opensips.org>; Kosmas Palios 
>> <kosmas.palios at gmail.com> <mailto:kosmas.palios at gmail.com>
>> *Subject:* Re: [OpenSIPS-Users] Call center issue in 3.1; call center 
>> not working after migration to 3.2
>>
>> 	
>> Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από 
>> bogdan at opensips.org <mailto:bogdan at opensips.org>. Μάθετε γιατί είναι 
>> σημαντικό <https://aka.ms/LearnAboutSenderIdentification>
>> 	
>>
>> Hi Kosmas,
>>
>> For Issue 1, try placing a t_newtran(); before calling the cc 
>> function in the script - this will prevent the issues due to 
>> retransmissions.
>>
>> Regards,
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>    https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>>    https://www.siphub.com  <https://www.siphub.com>
>> On 7/4/23 1:15 PM, Kosmas Palios wrote:
>>>
>>> Hello community,
>>>
>>>
>>> Our team has managed to setup a SIP Trunk to forward calls to a 
>>> number of SIP clients, in opensips 3.1.16. We are using the call 
>>> center module, and it works fine for low traffic. We would like to 
>>> ask two separate but related questions.
>>>
>>>
>>> ISSUE 1
>>>
>>>
>>> We are using the call center module to forward calls to 100 SIP 
>>> agents, and it works well if traffic is relatively low (about 25 
>>> incoming calls per minute). However, when traffic is higher, i.e. up 
>>> to 60 incoming calls per minute, we see calls getting rejected 
>>> because of cc_handle_call() failing with error message:
>>>
>>>
>>> DBG:b2b_entities:server_new: It is a retransmission, drop
>>>
>>> ERROR:b2b_logic:b2b_process_scenario_init: failed to create new b2b 
>>> server instance
>>>
>>>
>>> Unfortunately, every time this happens, an agent's status gets stuck 
>>> to "incall" forever, even though no cc_calls row includes him. So 
>>> that agent is lost.
>>>
>>>
>>> We are running in UDP mode, using 6 UDP workers. I’m attaching the 
>>> configuration file as opensips_3_1_16.cfg
>>>
>>> I can share the whole setup if needed.
>>>
>>>
>>> ISSUE 2
>>>
>>>
>>> We decided to migrate to 3.2 after seeing the bugfix to b2b_clients 
>>> leak. When we got to migrating the call center, we read this 
>>> blogpost: 
>>> https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/ 
>>> <https://blog.opensips.org/2021/01/06/the-script-driven-sip-b2bua/>
>>>
>>>
>>> " When comes to the modules using the b2b_logicAPI (providing 
>>> features on top of the B2B engine), the only affected one is the 
>>> call_centermodule. The change is minor – the xml file controlling 
>>> the call queuing logic was removed, as not needed any more. 
>>> Otherwise, in terms of usage, it is exactly the same."
>>>
>>>
>>> However, when we removed the lines:
>>>
>>>
>>> modparam("b2b_logic_xml","script_scenario", 
>>> "/etc/opensips/scenario_callcenter.xml")
>>>
>>> modparam("call_center", "b2b_scenario", "call center")
>>>
>>>
>>> the call center started behaving weird: it created another invite to 
>>> the sip trunk, instead of creating the invite to the agent (the call 
>>> id was good, but the to uri was wrong). I can give detailed logs on 
>>> this, but I wouldn't want to make this email any bigger than it 
>>> already is. I’m also attaching the configuration file as 
>>> opensips_3_2_13.cfg
>>>
>>>
>>> To sum up, our questions are:
>>>
>>>
>>> 1. Any ideas on what the problem is with creating a new server 
>>> instance for high numbers of calls?
>>>
>>> 2. What's the recommended way to migrate the call center to version 
>>> 3.2 ? Can we find an example script-driven call center somewhere?
>>>
>>>
>>> Thank you in advance for your help!
>>>
>>> Best regards,
>>>
>>> Kosmas
>>>
>>>
>>> P.S.: about our team: we are a small team from Athens, Greece 
>>> integrating voice assistants on various platforms. Unfortunately we 
>>> missed the latest Opensips summit held last September.
>>>
>>>
>>>
>>> _______________________________________________
>>> 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 bogdan at opensips.org  Tue Sep 19 09:26:34 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 12:26:34 +0300
Subject: [OpenSIPS-Users] rest_post logs
In-Reply-To: <CANoA6_K8JYTjKHr_PuYh6_Pd8kdkeX90-RPmu_p7Mf0kTwmaPw@mail.gmail.com>
References: <CANoA6_+pO=UJwvMcbqop-_-eUBO5UkyyNXHp8LdNNuq8_j8TTA@mail.gmail.com>
 <8c4e361b-a28f-4bb8-2601-7c7c33ee2d46@opensips.org>
 <CANoA6_K8JYTjKHr_PuYh6_Pd8kdkeX90-RPmu_p7Mf0kTwmaPw@mail.gmail.com>
Message-ID: <7cb76878-438c-ced4-571f-930b5ebe83cb@opensips.org>

Thanks Alberto for the update here.

YEs, that was my suspicion also, that the logs are generated by the lib 
itself and not by the opensips code. We may check if the curl lib gives 
us any possibility to control its verbosity - could you open a Feature 
Request here https://github.com/OpenSIPS/opensips/issues - we use this 
but only for coding related reports, not for community help in using 
OpenSIPS.

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 12:14 PM, Alberto wrote:
> Hi,
>
> Thanks for your reply,
>
> I'm using v3.2.
> I realized that that's the output from libcurl, which I can turn off 
> by setting log_stdout and log_stderror to no.
>
> Thanks again
>
> p.s. Have you ever thought about a forum instead of the mailing list? 
> I would have gone back and answered/closed my own question, but I 
> didn't know if I should have done that via mail.
>
>
> On Tue, 19 Sept 2023 at 09:57, Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     HI Alberto,
>
>     What OpenSIPS version do you have?
>
>     And are those the only log lines you get ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>        https://www.siphub.com  <https://www.siphub.com>
>
>     On 9/1/23 4:43 PM, Alberto wrote:
>>     Hi,
>>
>>     Is there a way to turn off logs for rest_post requests?
>>
>>     This is my log settings:
>>     log_level=-1
>>     xlog_level=-1
>>     log_stdout=yes
>>     log_stderror=yes
>>     log_facility=LOG_LOCAL0
>>
>>     This is the type of output I want to suppress:
>>     Sep 01 13:31:48 opensips opensips[59740]: > POST
>>     /api/opensips/doit HTTP/1.1
>>     Sep 01 13:31:48 opensips opensips[59740]: Host: 127.0.0.1
>>     Sep 01 13:31:48 opensips opensips[59740]: Accept: */*
>>     Sep 01 13:31:48 opensips opensips[59740]: Content-Type:
>>     application/json
>>     Sep 01 13:31:48 opensips opensips[59740]: Content-Length: 156
>>     Sep 01 13:31:48 opensips opensips[59740]:
>>     Sep 01 13:31:48 opensips opensips[59740]: * upload completely
>>     sent off: 156 out of 156 bytes
>>     Sep 01 13:31:48 opensips opensips[59740]: * Mark bundle as not
>>     supporting multiuse
>>     Sep 01 13:31:48 opensips opensips[59740]: < HTTP/1.1 200 OK
>>     Sep 01 13:31:48 opensips opensips[59740]: < Server: nginx
>>     Sep 01 13:31:48 opensips opensips[59740]: < Date: Fri, 01 Sep
>>     2023 13:31:48 GMT
>>     Sep 01 13:31:48 opensips opensips[59740]: < Content-Type:
>>     application/json; charset=utf-8
>>     Sep 01 13:31:48 opensips opensips[59740]: < Content-Length: 360
>>     Sep 01 13:31:48 opensips opensips[59740]: < Connection: keep-alive
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Access-Control-Allow-Origin: *
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Content-Security-Policy: default-src 'self';base-uri
>>     'self';font-src 'self' https: data:;form-action
>>     'self';frame-ancestors 'self';img-src 'self' data:;object-src
>>     'none';script-src 'self';script-src-attr 'none';style-src 'self'
>>     https: 'unsafe-inline';upgrade-insecure-requests
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Cross-Origin-Opener-Policy: same-origin
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Cross-Origin-Resource-Policy: same-origin
>>     Sep 01 13:31:48 opensips opensips[59740]: < Origin-Agent-Cluster: ?1
>>     Sep 01 13:31:48 opensips opensips[59740]: < Referrer-Policy:
>>     no-referrer
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Strict-Transport-Security: max-age=15552000; includeSubDomains
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     X-Content-Type-Options: nosniff
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     X-DNS-Prefetch-Control: off
>>     Sep 01 13:31:48 opensips opensips[59740]: < X-Download-Options:
>>     noopen
>>     Sep 01 13:31:48 opensips opensips[59740]: < X-Frame-Options:
>>     SAMEORIGIN
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     X-Permitted-Cross-Domain-Policies: none
>>     Sep 01 13:31:48 opensips opensips[59740]: < X-XSS-Protection: 0
>>     Sep 01 13:31:48 opensips opensips[59740]: <
>>     Sep 01 13:31:48 opensips opensips[59740]: * Connection #0 to host
>>     127.0.0.1 left intact
>>
>>     Thanks
>>
>>     _______________________________________________
>>     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 bogdan at opensips.org  Tue Sep 19 09:30:48 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 19 Sep 2023 12:30:48 +0300
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS
	registrations
In-Reply-To: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
References: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
Message-ID: <1e279a15-ff97-019a-3e62-55d7eab5009b@opensips.org>

Hi Ray,

Do you use any TCP aliasing options in your cfg ?

Regards,

Bogdan-Andrei Iancu

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

On 9/2/23 3:17 AM, Ray Jackson wrote:
>
> Hi all,
>
> I'm facing a weird issue which I think is related to broken TCP socket 
> reuse logic where the wrong client is receiving incoming calls due to 
> the wrong socket being used for the incoming INVITE.
>
> The scenario is when I have 2 clients registering using TLS behind NAT 
> at the same Public IPv4 address and both clients are using the same 
> private port number.  So client 1 registers and the Via and contact 
> header looks like:
>
> Via: SIP/2.0/TLS 
> 192.168.42.162:5062;branch=z9hG4bK1409895926;rport;alias Contact: 
> <sip:201 at 192.168.42.162:5062;transport=tls>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4>"
>
> Client 2 registers from behind the same Public IPv4 address and the 
> Via and contact header looks like:
>
> Via: SIP/2.0/TLS 192.168.42.186:5062;branch=z9hG4bK-aff1f3b3 Contact: 
> <sip:202 at 192.168.42.186:5062;transport=tls>;expires=300
>
> The location table shows Client 1 received field of 103.212.1.2:5062 
> and Client 103.212.1.2:23456
>
> When a call comes in for Client 1 the location lookup seems to return 
> the correct 'received' address and port (e.g. 103.212.1.2:5062) and 
> all the logs indicate that this is where the SIP INVITE *should* be 
> going to (in the $du field).  However when you check the SIP traffic 
> it selects Client 2's socket and the traffic goes to port 23456 
> instead of 5062.
>
> I think this is related somehow to the TCP port reuse logic inside 
> Opensips.  My suspicion is that Opensips is looking at the Contact or 
> Via port number (which is the same for both client 1 and 2) and then 
> somehow mapping this to the wrong TCP received socket.
>
> Does anybody have any suggestions here?  Should I be fixing the NAT in 
> the Contact header (using fix_nated_contact).  I read somewhere that 
> you shouldn't rewrite the Contact header to avoid problems with 
> sending a different Contact URI to the client on calls.  Or is this 
> issue more related to the Via header and the TCP port reuse logic 
> looking at this port instead of the actual received port when choosing 
> the outgoing socket?
>
> FYI: I am using both force_rport() and fix_nated_register() for 
> incoming registrations from these clients and matching_mode of 0 in 
> usrloc.  However, I am not using fix_nated_contact() for registrations.
>
> Thanks,
>
> Ray
>
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From kosmas.palios at athenarc.gr  Tue Sep 19 09:39:14 2023
From: kosmas.palios at athenarc.gr (Kosmas Palios)
Date: Tue, 19 Sep 2023 09:39:14 +0000
Subject: [OpenSIPS-Users] opensips deployment as load balancer for T.38
 traffic
In-Reply-To: <3c4a02cd-36be-5800-fe25-2b47ecd5b1a3@opensips.org>
References: <CAOL0t++ASQy=YDWJUcLbSafbsgC66TLdJ8LYMcEp2Ui6Dxtfug@mail.gmail.com>
 <d758442b-e058-b763-bf95-531d182abd3c@opensips.org>
 <DBBPR05MB65065A5E2B70407B0F78F32CE9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>
 <3c4a02cd-36be-5800-fe25-2b47ecd5b1a3@opensips.org>
Message-ID: <DBBPR05MB650670A49C86B3BB2D85B2A3E9FAA@DBBPR05MB6506.eurprd05.prod.outlook.com>

My apologies for responding to the wrong thread. Please ignore my last message, starting with:

"Thank you for your response.

We have tried generating ..."

Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Bogdan-Andrei Iancu <bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 12:13:51 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>; Kosmas Palios <kosmas.palios at athenarc.gr>; Tahir Almas Dhesi <tahir at ictinnovations.com>
Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic

Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>
Not sure what the callcenter has to do here with the LB scenario. LB is simply distributing (according to the load) a call to a bunch of similar back servers.

Regards,

Bogdan-Andrei Iancu

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

On 9/19/23 11:58 AM, Kosmas Palios wrote:
Thank you for your response.

We have tried generating different configs. We suspect that the problem is in the call center scenario (possibly the absence of). There seems to be a problem in connecting the incoming call to a new client instance.

The b2b mod generated INVITE that has the right agent in the INVITE Header but the wrong <To> Header. We have looked into the debug messages and there seem to be no reason for this issue. I will post some useful debug lines here later.

Sent from Outlook for iOS<https://aka.ms/o0ukef>
________________________________
From: Users <users-bounces at lists.opensips.org><mailto:users-bounces at lists.opensips.org> on behalf of Bogdan-Andrei Iancu <bogdan at opensips.org><mailto:bogdan at opensips.org>
Sent: Tuesday, September 19, 2023 11:46:01 AM
To: OpenSIPS users mailling list <users at lists.opensips.org><mailto:users at lists.opensips.org>; Tahir Almas Dhesi <tahir at ictinnovations.com><mailto:tahir at ictinnovations.com>
Subject: Re: [OpenSIPS-Users] opensips deployment as load balancer for T.38 traffic


Δεν λαμβάνετε συχνά μηνύματα ηλεκτρονικού ταχυδρομείου από bogdan at opensips.org<mailto:bogdan at opensips.org>. Μάθετε γιατί είναι σημαντικό<https://aka.ms/LearnAboutSenderIdentification>

Hi,

Take a look here, it might be a good starting point:
    https://www.opensips.org/Documentation/Generating-Configs-3-2

Regards,

Bogdan-Andrei Iancu

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

On 9/5/23 3:22 PM, Tahir Almas Dhesi wrote:
Interested to know how we can deploy opensips as load balancer for  T.38 calls , whether it will work fine or we need to use rabbitmq for load balancing

scenario is

ICTFax => opesips load balancer => freeswitch nodes



regards
Tahir Almas

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT




_______________________________________________
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 prathibhab.tvm at gmail.com  Tue Sep 19 10:33:10 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 19 Sep 2023 16:03:10 +0530
Subject: [OpenSIPS-Users] ERROR:rtpengine:rtpe_function_call: proxy
 replied with error: Unknown call-id
In-Reply-To: <CAJh_-cAZodsOAToQEcA+mPowsSgK09tgYZ2KqpidugAe3x3nBQ@mail.gmail.com>
References: <CAJh_-cBEV7Bgx_2Ct6A_Qm1S9+pgSXhvsRv4JcEdUgwL6dTtvA@mail.gmail.com>
 <CAJh_-cAZodsOAToQEcA+mPowsSgK09tgYZ2KqpidugAe3x3nBQ@mail.gmail.com>
Message-ID: <CAJh_-cCm8Er4T4zfH+aNG+noS7sf25dMCjkJdB4MZku3noSaxw@mail.gmail.com>

Trying to bind the socket for port = '36412'
rtpengine[7938]: DEBUG: [2fanirs85upoeuj766as]: [core] Something already
keeps this port, trying to take another port(s)

On Tue, 19 Sept 2023 at 14:38, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> Can anyone pls help?
>
> On Tue, 19 Sept 2023 at 13:39, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> I am getting this error:
>>
>> ERROR:rtpengine:rtpe_function_call: proxy replied with error: Unknown
>> call-id
>>
>> How to resolve this?
>>
>> I've attached the opensips.cf file with this email.
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From mickael at winlux.fr  Wed Sep 20 10:07:12 2023
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 20 Sep 2023 12:07:12 +0200
Subject: [OpenSIPS-Users] Catch rtp ip and port for each side
Message-ID: <CA+fRVSXPh5k1d9_E-NsCcOicBY=uvwtkg100XHKqW+hxHrw=QQ@mail.gmail.com>

Hi all,
for an specific application, I need to get rtp ip and port for each side
(after 200OK)
can you tell me, if there is a way to catch rtp IP and port from sdp for
each side without regex ?

I can use this line $var(aline) = $(rb{sdp.line,c,0}); + regex, But maybe
there exists an easy way, ex:  variables with rtp ip and port for each side.

thanks in advance
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From bogdan at opensips.org  Wed Sep 20 11:14:30 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 20 Sep 2023 14:14:30 +0300
Subject: [OpenSIPS-Users] Catch rtp ip and port for each side
In-Reply-To: <CA+fRVSXPh5k1d9_E-NsCcOicBY=uvwtkg100XHKqW+hxHrw=QQ@mail.gmail.com>
References: <CA+fRVSXPh5k1d9_E-NsCcOicBY=uvwtkg100XHKqW+hxHrw=QQ@mail.gmail.com>
Message-ID: <2463fa95-7230-baca-7c81-8d82a9335107@opensips.org>

Hi Mickael,

Unfortunately nothing simpler :(

Regards,

Bogdan-Andrei Iancu

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

On 9/20/23 1:07 PM, Mickael Hubert wrote:
> Hi all,
> for an specific application, I need to get rtp ip and port for each 
> side (after 200OK)
> can you tell me, if there is a way to catch rtp IP and port from sdp 
> for each side without regex ?
>
> I can use this line $var(aline) = $(rb{sdp.line,c,0}); + regex, But 
> maybe there exists an easy way, ex:  variables with rtp ip and port 
> for each side.
>
> thanks in advance
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Wed Sep 20 11:19:48 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 20 Sep 2023 14:19:48 +0300
Subject: [OpenSIPS-Users] Compatibility Inquiry: Opensips 3.2.12 with
 RHEL 9
In-Reply-To: <CAPHmyfxsq63yPW8DX98=EYpP+cp+S9b8_EHuON-FtpSWTbzddw@mail.gmail.com>
References: <CAPHmyfxsq63yPW8DX98=EYpP+cp+S9b8_EHuON-FtpSWTbzddw@mail.gmail.com>
Message-ID: <e7b21512-c24b-e44d-cf23-48914c95a38c@opensips.org>

Hi,

OpenSIPS does not have any particular affinities to certain OS or 
version - it depends only on certain libraries + version. So, give it a 
try and if you encounter issues, just report.

Regards,

Bogdan-Andrei Iancu

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

On 8/23/23 2:38 PM, Vinayak Makwana via Users wrote:
> Hello all,
>
>        I hope this message finds you well. I am currently in the 
> process of installing Opensips 3.2.12 Version on RHEL 9 OS. However, 
> I've encountered some dependency issues during the compilation of the 
> required modules. To address this, I am actively working on resolving 
> these issues and installing the necessary dependencies by compiling 
> them from their respective source codes.
>
>         My primary concern is to determine the compatibility of 
> Opensips 3.2.12 with the RHEL 9 Operating System. I would greatly 
> appreciate any insights or guidance you can provide on this matter. 
> Thank you for your time and assistance.
>
> Regards,
> Vinayak Makwana
>
> *Disclaimer*
> In addition to generic Disclaimer which you have agreed on our 
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> or otherwise) arising from any third party taking any action, or 
> refraining from taking any action on the basis of any of the 
> information contained in this email is hereby excluded.
>
> *Confidentiality*
> This communication (including any attachment/s) is intended only for 
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> _______________________________________________
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From bogdan at opensips.org  Wed Sep 20 11:22:40 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 20 Sep 2023 14:22:40 +0300
Subject: [OpenSIPS-Users] handing To URI params with uac_replace_to()
In-Reply-To: <CH2PR12MB42629780B549513B3F1FD8D3A90DA@CH2PR12MB4262.namprd12.prod.outlook.com>
References: <CH2PR12MB42629780B549513B3F1FD8D3A90DA@CH2PR12MB4262.namprd12.prod.outlook.com>
Message-ID: <be79211c-dfc7-1a55-e152-198ba2804d29@opensips.org>

Hi Jeff,

the UAC module does not offer any URI param masquerading, only for the 
display and URI part.

The only option you have is to do some textops to remove and add that 
param, manually, from script.

Regards,

Bogdan-Andrei Iancu

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

On 8/8/23 5:17 PM, Pyle, Jeff wrote:
> Hello,
>
> This is on OpenSIPS 3.2.13
>
> I am running uac_replace_to() in branch_route, with restore_mode on 
> 'auto'. It works as expected...almost.
>
> The b-leg attaches To URI params in addition to the tag on its 
> replies. I would like to strip these params before relaying back to 
> the a-leg. Is that possible?
>
> Then, of course, the tags would be restored when sending messages back 
> towards the b-leg, as a function of the 'auto' mode.
>
>
>
> Regards,
> Jeff
>
>
>
> This message is subject to Fusion Connect, Inc.’s email communication 
> policy: www.fusionconnect.com/email-policy
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From prathibhab.tvm at gmail.com  Wed Sep 20 11:32:21 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Wed, 20 Sep 2023 17:02:21 +0530
Subject: [OpenSIPS-Users] Connection not getting established
Message-ID: <CAJh_-cAhPL1rAKZrOPOx_Zkt5oExea=LZPqwvaSN7rrjreZk1Q@mail.gmail.com>

I changed /etc/sysconfig/rtpengine as

OPTIONS="-n 172.31.34.24:2225 -m 10000 -M 20000 -L 7 --log-facility=local0
--table=-1 --delete-delay=0 --timeout=60 --silent-timeout=600
--final-timeout=7200 –offer-timeout=60 --num-threads=12 --tos=184
–no-fallback auto-bridge"

In /var/log/syslog it is showing the following error:

*Setting ICE candidate pair 1Go6nTJGQOlRawIz:3091388503:1 as failed*

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Wed Sep 20 11:57:05 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Wed, 20 Sep 2023 17:27:05 +0530
Subject: [OpenSIPS-Users] Connection not getting established
In-Reply-To: <CAJh_-cAhPL1rAKZrOPOx_Zkt5oExea=LZPqwvaSN7rrjreZk1Q@mail.gmail.com>
References: <CAJh_-cAhPL1rAKZrOPOx_Zkt5oExea=LZPqwvaSN7rrjreZk1Q@mail.gmail.com>
Message-ID: <CAJh_-cC68GMo64Z+Gst8yOuNLzHjvsvSochNzc1gmP+chP73Gg@mail.gmail.com>

I am also getting these error w.r.t opensips in syslog

ERROR:core:tcp_connect_blocking_timeout: connect timed out, 299361 us
elapsed out of 300000 us
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:proto_wss:ws_connect: connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:proto_wss:proto_wss_send: connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:tm:msg_send: send() to 14.139.183.221:62420 for proto wss/6 failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:tm:t_forward_nonack: sending request failed

On Wed, 20 Sept 2023 at 17:02, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I changed /etc/sysconfig/rtpengine as
>
> OPTIONS="-n 172.31.34.24:2225 -m 10000 -M 20000 -L 7
> --log-facility=local0 --table=-1 --delete-delay=0 --timeout=60
> --silent-timeout=600 --final-timeout=7200 –offer-timeout=60
> --num-threads=12 --tos=184 –no-fallback auto-bridge"
>
> In /var/log/syslog it is showing the following error:
>
> *Setting ICE candidate pair 1Go6nTJGQOlRawIz:3091388503:1 as failed*
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From ray at hero.co.nz  Fri Sep 22 02:03:51 2023
From: ray at hero.co.nz (Ray Jackson)
Date: Fri, 22 Sep 2023 14:03:51 +1200
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS
	registrations
In-Reply-To: <1e279a15-ff97-019a-3e62-55d7eab5009b@opensips.org>
References: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
 <1e279a15-ff97-019a-3e62-55d7eab5009b@opensips.org>
Message-ID: <d1ee08b6-5545-1dbb-6423-a8241a07d100@hero.co.nz>

Hi Bogdan,

Yes, we have the following enabled in our config:

tcp_accept_aliases=1

I assume this is the culprit then and we are inadvertently sending calls 
down the wrong TCP socket here to the wrong user due to this being 
enabled?  This is quite a nasty setting to have enabled when we are 
dealing with CGNAT'd customers who are sharing public IP addresses but 
are completely unrelated users!

I will disable this setting and see if that clears up the issue for us.  
We have in fact had another case just today of the same issue happening 
(User A is receiving User B's incoming calls!)

Thanks for highlighting this and let me know if there is anything else I 
should look at in our config.

Thanks,

Ray

On 19/09/23 9:30 pm, Bogdan-Andrei Iancu wrote:

> Hi Ray,
>
> Do you use any TCP aliasing options in your cfg ?
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
>    https://www.siphub.com
> On 9/2/23 3:17 AM, Ray Jackson wrote:
>>
>> Hi all,
>>
>> I'm facing a weird issue which I think is related to broken TCP 
>> socket reuse logic where the wrong client is receiving incoming calls 
>> due to the wrong socket being used for the incoming INVITE.
>>
>> The scenario is when I have 2 clients registering using TLS behind 
>> NAT at the same Public IPv4 address and both clients are using the 
>> same private port number.  So client 1 registers and the Via and 
>> contact header looks like:
>>
>> Via: SIP/2.0/TLS 
>> 192.168.42.162:5062;branch=z9hG4bK1409895926;rport;alias Contact: 
>> <sip:201 at 192.168.42.162:5062;transport=tls>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4>"
>>
>> Client 2 registers from behind the same Public IPv4 address and the 
>> Via and contact header looks like:
>>
>> Via: SIP/2.0/TLS 192.168.42.186:5062;branch=z9hG4bK-aff1f3b3 Contact: 
>> <sip:202 at 192.168.42.186:5062;transport=tls>;expires=300
>>
>> The location table shows Client 1 received field of 103.212.1.2:5062 
>> and Client 103.212.1.2:23456
>>
>> When a call comes in for Client 1 the location lookup seems to return 
>> the correct 'received' address and port (e.g. 103.212.1.2:5062) and 
>> all the logs indicate that this is where the SIP INVITE *should* be 
>> going to (in the $du field). However when you check the SIP traffic 
>> it selects Client 2's socket and the traffic goes to port 23456 
>> instead of 5062.
>>
>> I think this is related somehow to the TCP port reuse logic inside 
>> Opensips.  My suspicion is that Opensips is looking at the Contact or 
>> Via port number (which is the same for both client 1 and 2) and then 
>> somehow mapping this to the wrong TCP received socket.
>>
>> Does anybody have any suggestions here?  Should I be fixing the NAT 
>> in the Contact header (using fix_nated_contact).  I read somewhere 
>> that you shouldn't rewrite the Contact header to avoid problems with 
>> sending a different Contact URI to the client on calls.  Or is this 
>> issue more related to the Via header and the TCP port reuse logic 
>> looking at this port instead of the actual received port when 
>> choosing the outgoing socket?
>>
>> FYI: I am using both force_rport() and fix_nated_register() for 
>> incoming registrations from these clients and matching_mode of 0 in 
>> usrloc.  However, I am not using fix_nated_contact() for registrations.
>>
>> Thanks,
>>
>> Ray
>>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From prathibhab.tvm at gmail.com  Sat Sep 23 03:49:57 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 09:19:57 +0530
Subject: [OpenSIPS-Users] nathelper script bug
Message-ID: <CAJh_-cBB1nJxuqtms9UQNpHMDcUJ8SJAUNWW_o2ghNkiiGDvyg@mail.gmail.com>

ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change
URI Contact

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 04:11:12 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 09:41:12 +0530
Subject: [OpenSIPS-Users] nathelper script bug
In-Reply-To: <CAJh_-cBB1nJxuqtms9UQNpHMDcUJ8SJAUNWW_o2ghNkiiGDvyg@mail.gmail.com>
References: <CAJh_-cBB1nJxuqtms9UQNpHMDcUJ8SJAUNWW_o2ghNkiiGDvyg@mail.gmail.com>
Message-ID: <CAJh_-cAf72s0c9i_gVj0cFMmL1mGx_qwb0nBrpbbN=NBPj+5yA@mail.gmail.com>

I've used fix_nated_contact() in the following places in opensips.cfg file:

route
if has_to_tag()


On Sat, 23 Sept 2023 at 09:19, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> ERROR:nathelper:fix_nated_contact_f: SCRIPT BUG - second attempt to change
> URI Contact
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 12:55:52 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 18:25:52 +0530
Subject: [OpenSIPS-Users] connection not getting established
Message-ID: <CAJh_-cAGpGkOsubvqDBb5HuAWBKhTfwgpRoSNkzPOYeOkJwfYA@mail.gmail.com>

ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:proto_wss:ws_connect: connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR:proto_wss:proto_wss_send: connect failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]: ERROR™️msg_send:
send() to 14.139.183.221:62420 for proto wss/6 failed
Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
ERROR™️t_forward_nonack: sending request failed

-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 12:56:45 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 18:26:45 +0530
Subject: [OpenSIPS-Users] connection not getting established
In-Reply-To: <CAJh_-cAGpGkOsubvqDBb5HuAWBKhTfwgpRoSNkzPOYeOkJwfYA@mail.gmail.com>
References: <CAJh_-cAGpGkOsubvqDBb5HuAWBKhTfwgpRoSNkzPOYeOkJwfYA@mail.gmail.com>
Message-ID: <CAJh_-cCyTzDVoqtzx38NteFSHgFufmz6pK=f4SLwMJX8bQwsiA@mail.gmail.com>

I am using opensips-3.3 I am unable to resolve the issue. Pls help.


On Sat, 23 Sept 2023 at 18:25, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed
> Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
> ERROR:proto_wss:ws_connect: connect failed
> Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
> ERROR:proto_wss:proto_wss_send: connect failed
> Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
> ERROR™️msg_send: send() to 14.139.183.221:62420 for proto wss/6 failed
> Sep 20 11:45:39 ip-172-31-34-24 /usr/sbin/opensips[42407]:
> ERROR™️t_forward_nonack: sending request failed
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 14:58:45 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 20:28:45 +0530
Subject: [OpenSIPS-Users] Too many Hops
Message-ID: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>

While executing opensips using sipml5, I'm getting Too Mnay Hops error.
This occurs when I am using ip address instead of domain name for the
websocket url. How to resolve this?

-- 
Regards,
B.Prathibha
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From david.villasmil.work at gmail.com  Sat Sep 23 15:06:55 2023
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sat, 23 Sep 2023 08:06:55 -0700
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
Message-ID: <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>

Make you opensips finds a next hope. This usually happens when no next hope
is set and opensips forwards to itself.

On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> While executing opensips using sipml5, I'm getting Too Mnay Hops error.
> This occurs when I am using ip address instead of domain name for the
> websocket url. How to resolve this?
>
>
> --
> Regards,
> B.Prathibha
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From prathibhab.tvm at gmail.com  Sat Sep 23 15:23:09 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 20:53:09 +0530
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
 <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
Message-ID: <CAJh_-cCJsiaw1rRsxDLaJEU30A2xpJ4axv3t+chwcjisYuoKwA@mail.gmail.com>

I've set public ip as alias address in opensips.cfg and the issue is
resolved. Now I'm getting these errors:
ERROR:tls_openssl:openssl_tls_write: TLS connection to 42.111.162.67:31273
write failed (5:-1:9)
Sep 23 15:20:13 ip-172-31-34-24 /usr/sbin/opensips[3690]:
ERROR:tls_openssl:openssl_tls_write: TLS write error:
Sep 23 15:20:13 ip-172-31-34-24 /usr/sbin/opensips[3690]:
ERROR:tls_openssl:openssl_tls_blocking_write: TLS failed to send data
Sep 23 15:20:13 ip-172-31-34-24 /usr/sbin/opensips[3690]:
ERROR:tls_openssl:openssl_tls_write: TLS connection to 111.92.8.58:52260
write failed (5:-1:9)
Sep 23 15:20:13 ip-172-31-34-24 /usr/sbin/opensips[3690]:
ERROR:tls_openssl:openssl_tls_write: TLS write error:
Sep 23 15:20:13 ip-172-31-34-24 /usr/sbin/opensips[3690]:
ERROR:tls_openssl:openssl_tls_blocking_write: TLS failed to send data

How to resolve these issues?

On Sat, 23 Sept 2023 at 20:39, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Make you opensips finds a next hope. This usually happens when no next
> hope is set and opensips forwards to itself.
>
> On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> While executing opensips using sipml5, I'm getting Too Mnay Hops error.
>> This occurs when I am using ip address instead of domain name for the
>> websocket url. How to resolve this?
>>
>>
>> --
>> Regards,
>> B.Prathibha
>> _______________________________________________
>> 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,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 15:24:45 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 20:54:45 +0530
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
 <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
Message-ID: <CAJh_-cCLFuTeX4PzhcJCvhqastRLXLR3=h+Z_THimH=4UWdLYw@mail.gmail.com>

socket=udp:172.31.34.24:5060 as 43.205.195.119:5060
socket=tcp:172.31.34.24:5060 as 43.205.195.119:5060
socket=tls:172.31.34.24:5061 as 43.205.195.119:5061
socket=wss:172.31.34.24:7443 as 43.205.195.119:7443

On Sat, 23 Sept 2023 at 20:39, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Make you opensips finds a next hope. This usually happens when no next
> hope is set and opensips forwards to itself.
>
> On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> While executing opensips using sipml5, I'm getting Too Mnay Hops error.
>> This occurs when I am using ip address instead of domain name for the
>> websocket url. How to resolve this?
>>
>>
>> --
>> Regards,
>> B.Prathibha
>> _______________________________________________
>> 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,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 15:29:08 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 20:59:08 +0530
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAJh_-cCLFuTeX4PzhcJCvhqastRLXLR3=h+Z_THimH=4UWdLYw@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
 <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
 <CAJh_-cCLFuTeX4PzhcJCvhqastRLXLR3=h+Z_THimH=4UWdLYw@mail.gmail.com>
Message-ID: <CAJh_-cCYoX5umkoG8-Q+n5oJq=jTd6FrsyD4pamEg60yeP5GjQ@mail.gmail.com>

How to use self signed certificate in opensips? I am getting this error:

SSL routines::sslv3 alert certificate unknown

On Sat, 23 Sept 2023 at 20:54, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> socket=udp:172.31.34.24:5060 as 43.205.195.119:5060
> socket=tcp:172.31.34.24:5060 as 43.205.195.119:5060
> socket=tls:172.31.34.24:5061 as 43.205.195.119:5061
> socket=wss:172.31.34.24:7443 as 43.205.195.119:7443
>
> On Sat, 23 Sept 2023 at 20:39, David Villasmil <
> david.villasmil.work at gmail.com> wrote:
>
>> Make you opensips finds a next hope. This usually happens when no next
>> hope is set and opensips forwards to itself.
>>
>> On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com>
>> wrote:
>>
>>> While executing opensips using sipml5, I'm getting Too Mnay Hops error.
>>> This occurs when I am using ip address instead of domain name for the
>>> websocket url. How to resolve this?
>>>
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>> _______________________________________________
>>> 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,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 15:42:51 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 21:12:51 +0530
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAJh_-cCYoX5umkoG8-Q+n5oJq=jTd6FrsyD4pamEg60yeP5GjQ@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
 <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
 <CAJh_-cCLFuTeX4PzhcJCvhqastRLXLR3=h+Z_THimH=4UWdLYw@mail.gmail.com>
 <CAJh_-cCYoX5umkoG8-Q+n5oJq=jTd6FrsyD4pamEg60yeP5GjQ@mail.gmail.com>
Message-ID: <CAJh_-cDbb9za0WXZ-LjHgRee7f8MqR9qGedB86se_yAC72iqEA@mail.gmail.com>

ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
42.111.162.67:31290 failed to accept
Sep 23 15:41:10 ip-172-31-34-24 /usr/sbin/opensips[4959]:
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A000416:SSL
routines::sslv3 alert certificate unknown
Sep 23 15:41:10 ip-172-31-34-24 /usr/sbin/opensips[4959]:
ERROR:proto_wss:wss_read_req: cannot fix read connection
Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
111.92.8.58:52695 failed to accept
Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A000416:SSL
routines::sslv3 alert certificate unknown
Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
ERROR:proto_wss:wss_read_req: cannot fix read connection

On Sat, 23 Sept 2023 at 20:59, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> How to use self signed certificate in opensips? I am getting this error:
>
> SSL routines::sslv3 alert certificate unknown
>
> On Sat, 23 Sept 2023 at 20:54, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> socket=udp:172.31.34.24:5060 as 43.205.195.119:5060
>> socket=tcp:172.31.34.24:5060 as 43.205.195.119:5060
>> socket=tls:172.31.34.24:5061 as 43.205.195.119:5061
>> socket=wss:172.31.34.24:7443 as 43.205.195.119:7443
>>
>> On Sat, 23 Sept 2023 at 20:39, David Villasmil <
>> david.villasmil.work at gmail.com> wrote:
>>
>>> Make you opensips finds a next hope. This usually happens when no next
>>> hope is set and opensips forwards to itself.
>>>
>>> On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com>
>>> wrote:
>>>
>>>> While executing opensips using sipml5, I'm getting Too Mnay Hops error.
>>>> This occurs when I am using ip address instead of domain name for the
>>>> websocket url. How to resolve this?
>>>>
>>>>
>>>> --
>>>> Regards,
>>>> B.Prathibha
>>>> _______________________________________________
>>>> 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,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sat Sep 23 15:46:03 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sat, 23 Sep 2023 21:16:03 +0530
Subject: [OpenSIPS-Users] Too many Hops
In-Reply-To: <CAJh_-cDbb9za0WXZ-LjHgRee7f8MqR9qGedB86se_yAC72iqEA@mail.gmail.com>
References: <CAJh_-cDMXt6AtR6SHKjozTGEriUYVAdwPzmp3mHV=-tDvregjw@mail.gmail.com>
 <CAFGRPVr19q8Jz5cJdhY14mWUhx0VcYOnNMFVBo5_D3pDOH-DVw@mail.gmail.com>
 <CAJh_-cCLFuTeX4PzhcJCvhqastRLXLR3=h+Z_THimH=4UWdLYw@mail.gmail.com>
 <CAJh_-cCYoX5umkoG8-Q+n5oJq=jTd6FrsyD4pamEg60yeP5GjQ@mail.gmail.com>
 <CAJh_-cDbb9za0WXZ-LjHgRee7f8MqR9qGedB86se_yAC72iqEA@mail.gmail.com>
Message-ID: <CAJh_-cAx93iZhyuNbmNLt=Su58h-eE=Ah3tnbb5Ga5HYn=gaig@mail.gmail.com>

I'm using a certificate generated using lets encrypt.

On Sat, 23 Sept 2023 at 21:12, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
> 42.111.162.67:31290 failed to accept
> Sep 23 15:41:10 ip-172-31-34-24 /usr/sbin/opensips[4959]:
> ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A000416:SSL
> routines::sslv3 alert certificate unknown
> Sep 23 15:41:10 ip-172-31-34-24 /usr/sbin/opensips[4959]:
> ERROR:proto_wss:wss_read_req: cannot fix read connection
> Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
> ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
> 111.92.8.58:52695 failed to accept
> Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
> ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A000416:SSL
> routines::sslv3 alert certificate unknown
> Sep 23 15:41:11 ip-172-31-34-24 /usr/sbin/opensips[4959]:
> ERROR:proto_wss:wss_read_req: cannot fix read connection
>
> On Sat, 23 Sept 2023 at 20:59, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> How to use self signed certificate in opensips? I am getting this error:
>>
>> SSL routines::sslv3 alert certificate unknown
>>
>> On Sat, 23 Sept 2023 at 20:54, Prathibha B <prathibhab.tvm at gmail.com>
>> wrote:
>>
>>> socket=udp:172.31.34.24:5060 as 43.205.195.119:5060
>>> socket=tcp:172.31.34.24:5060 as 43.205.195.119:5060
>>> socket=tls:172.31.34.24:5061 as 43.205.195.119:5061
>>> socket=wss:172.31.34.24:7443 as 43.205.195.119:7443
>>>
>>> On Sat, 23 Sept 2023 at 20:39, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Make you opensips finds a next hope. This usually happens when no next
>>>> hope is set and opensips forwards to itself.
>>>>
>>>> On Sat, 23 Sep 2023 at 07:59, Prathibha B <prathibhab.tvm at gmail.com>
>>>> wrote:
>>>>
>>>>> While executing opensips using sipml5, I'm getting Too Mnay Hops
>>>>> error. This occurs when I am using ip address instead of domain name for
>>>>> the websocket url. How to resolve this?
>>>>>
>>>>>
>>>>> --
>>>>> Regards,
>>>>> B.Prathibha
>>>>> _______________________________________________
>>>>> 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,
>>> B.Prathibha
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Sun Sep 24 05:34:34 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Sun, 24 Sep 2023 11:04:34 +0530
Subject: [OpenSIPS-Users] SSL Certificate error
Message-ID: <CAJh_-cA3HXMZi4C_v=gbR07+C97vsVfNTzvTSxZZz79HxoEWgA@mail.gmail.com>

TLSv1.2 (OUT), TLS header, Supplemental data (23):
* OpenSSL SSL_write: Connection reset by peer, errno 104
* Failed sending HTTP request
* Connection #0 to host bp3.erss.in left intact
curl: (55) OpenSSL SSL_write: Connection reset by peer, errno 104

-- 
Regards,
B.Prathibha
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From vincent.horst at ehvc.nl  Mon Sep 25 20:30:26 2023
From: vincent.horst at ehvc.nl (Vincent Horst)
Date: Mon, 25 Sep 2023 22:30:26 +0200
Subject: [OpenSIPS-Users] Opensips 3.4 Not sending hep to homer server
Message-ID: <CAK1AzRXtMTe8z-R6GxRkO1PsmCYa8Aqq=0x1otrwwxg-xtx2gw@mail.gmail.com>

Hi All,

I have a issue where opensips is not sending the hep files towards homer.
I have used a manual and have add below to the .cfg file.

Sockets: (It is unclear for my why I should put here the local adres, when
I put the remote IP of homer, I get an error in the config:
socket=hep_udp:127.0.0.1:6061
socket=hep_tcp:127.0.0.1:6061
If I remove them from the config the hep module gives an error.

Second part in the .cfg:
### Configure an HEP Endpoint
loadmodule "proto_hep.so"
modparam("proto_hep", "hep_id", "[hid]10.0.0.216:9060
;transport=udp;version=3")
#### Configure Tracer module to use the HEP Protocol instance id [hid]
loadmodule "tracer.so"
modparam("tracer", "trace_on", 1)
modparam("tracer", "trace_id", "[tid]uri=hep:hid")

And then the third part,
I noticed that it make no different when I put it above in the route logic
or below.

route[to_homer] {
$var(trace_id) = "tid";
if (!has_totag()) {
if (is_method("INVITE")) { $var(trace_type) = "dialog"; }
else if (!is_method("CANCEL")) { $var(trace_type) = "transaction"; }
} else { $var(trace_type) = NULL; }
switch ($var(trace_type)) {
case "dialog":
trace("$var(trace_id)", "d", "sip|xlog|rest");
break;
case "transaction":
trace("$var(trace_id)", "t", "sip|xlog");
break;
}
}

I'm not getting an error, but I'm also not seeing any packets received on
the homer server.
Can someone point me in the right direction or explain how I can add
details to trouble shoot?

Cheers,
Vincent
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From mcrans at gmail.com  Tue Sep 26 06:21:29 2023
From: mcrans at gmail.com (Michel crans)
Date: Tue, 26 Sep 2023 08:21:29 +0200
Subject: [OpenSIPS-Users] Opensips 3.4 Not sending hep to homer server
In-Reply-To: <CAK1AzRXtMTe8z-R6GxRkO1PsmCYa8Aqq=0x1otrwwxg-xtx2gw@mail.gmail.com>
References: <CAK1AzRXtMTe8z-R6GxRkO1PsmCYa8Aqq=0x1otrwwxg-xtx2gw@mail.gmail.com>
Message-ID: <CAPWMwMF1AbxfWJ8cx8rcuH=S4NWTauaCCvDXT3RB73Wow8XYRQ@mail.gmail.com>

If you're not seeing any packets received on the Homer server, there could
be several reasons for this issue. Let's go through some troubleshooting
steps:

   1.

   Check Network Connectivity:
   - Ensure that there is network connectivity between your OpenSIPS server
      and the Homer server. You should be able to ping the Homer
server from the
      OpenSIPS server.
   2.

   Homer Server Configuration:
   - Verify that your Homer server is correctly configured to receive HEP
      packets on the specified IP and port (in this case, 127.0.0.1:6061).
      Check Homer's logs for any incoming connection attempts or errors.
   3.

   Socket Configuration:
   - It seems you are using 127.0.0.1 as the IP address in your socket
      configuration (socket=hep_udp:127.0.0.1:6061). This means OpenSIPS
      will send HEP packets to localhost (the same server it's running on).
      Ensure that this is the correct configuration. If Homer is on a different
      server, you should use the Homer server's IP address instead.
   4.

   Firewall and Security Rules:
   - Make sure that there are no firewall rules or security policies
      blocking traffic on the specified HEP port (6061) or the network
interface
      being used for communication.
   5.

   Logging and Debugging:
   - Enable verbose logging in OpenSIPS to see if there are any error
      messages related to HEP. You can do this by setting the log_level
      parameter in your OpenSIPS configuration file.

   cfgCopy code
   modparam("log", "log_level", 3)

   This will increase the logging verbosity, and you can check the OpenSIPS
   logs for any relevant error messages.
   6.

   Verify Route Execution:
   - Ensure that your route[to_homer] block is being executed when you
      expect it to. You can add additional log messages in this block to verify
      that it's being triggered.
   7.

   Check HEP Configuration:
   - Review your HEP configuration parameters, such as hep_id and trace_id,
      to ensure they are correctly set. Make sure they match the
configuration on
      the Homer server.
   8.

   Firewall on Homer Server:
   - Check if there is any firewall or security software running on the
      Homer server that might be blocking incoming HEP packets.
   9.

   Homer Server Logs:
   - Examine the logs on your Homer server to see if there are any error
      messages or indications of failed connections.
   10.

   Packet Capture:
   - You can use tools like Wireshark to capture network traffic on the
      OpenSIPS server and check if the HEP packets are actually being
sent. This
      can help you confirm whether the issue is with OpenSIPS or the network.

By systematically going through these troubleshooting steps, you should be
able to identify the root cause of why OpenSIPS is not sending HEP packets
to your Homer server. Remember to make configuration adjustments as needed
to match your specific network setup and requirements.

Op ma 25 sep 2023 om 22:34 schreef Vincent Horst via Users <
users at lists.opensips.org>:

> Hi All,
>
> I have a issue where opensips is not sending the hep files towards homer.
> I have used a manual and have add below to the .cfg file.
>
> Sockets: (It is unclear for my why I should put here the local adres, when
> I put the remote IP of homer, I get an error in the config:
> socket=hep_udp:127.0.0.1:6061
> socket=hep_tcp:127.0.0.1:6061
> If I remove them from the config the hep module gives an error.
>
> Second part in the .cfg:
> ### Configure an HEP Endpoint
> loadmodule "proto_hep.so"
> modparam("proto_hep", "hep_id", "[hid]10.0.0.216:9060
> ;transport=udp;version=3")
> #### Configure Tracer module to use the HEP Protocol instance id [hid]
> loadmodule "tracer.so"
> modparam("tracer", "trace_on", 1)
> modparam("tracer", "trace_id", "[tid]uri=hep:hid")
>
> And then the third part,
> I noticed that it make no different when I put it above in the route logic
> or below.
>
> route[to_homer] {
> $var(trace_id) = "tid";
> if (!has_totag()) {
> if (is_method("INVITE")) { $var(trace_type) = "dialog"; }
> else if (!is_method("CANCEL")) { $var(trace_type) = "transaction"; }
> } else { $var(trace_type) = NULL; }
> switch ($var(trace_type)) {
> case "dialog":
> trace("$var(trace_id)", "d", "sip|xlog|rest");
> break;
> case "transaction":
> trace("$var(trace_id)", "t", "sip|xlog");
> break;
> }
> }
>
> I'm not getting an error, but I'm also not seeing any packets received on
> the homer server.
> Can someone point me in the right direction or explain how I can add
> details to trouble shoot?
>
> Cheers,
> Vincent
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From prathibhab.tvm at gmail.com  Tue Sep 26 08:10:34 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 26 Sep 2023 13:40:34 +0530
Subject: [OpenSIPS-Users] Temporarily unavailable
Message-ID: <CAJh_-cBfD40+uUnYoc7GD7EOZBuDSydbJEu8GkgBGiYp4cFj4w@mail.gmail.com>

I've created two users and they are registered to opensips. When I try to
connect, user 1 to user 2, I get temporarily unavailable. Config file is
attached with this email.

-- 
Regards,
B.Prathibha
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#
# OpenSIPS residential configuration script
#     by OpenSIPS Solutions <team at opensips-solutions.com>
#
# Please refer to the Core CookBook at:
#      http://www.opensips.org/Resources/DocsCookbooks
# for a explanation of possible statements, functions and parameters.
#


####### Global Parameters #########

log_level=5
xlog_level=5
log_stderror=no
log_facility=LOG_LOCAL0

open_files_limit=4096

udp_workers=4

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes


socket=udp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tcp:172.31.34.24:5060 as 65.2.167.22:5060  # CUSTOMIZE ME
socket=tls:172.31.34.24:5061 as 65.2.167.22:5061  # CUSTOMIZE ME
socket=ws:172.31.34.24:8080 as 65.2.167.22:8080
socket=wss:172.31.34.24:7443 as 65.2.167.22:7443
####### Modules Section ########

#set module path
mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/"

#### SIGNALING module
loadmodule "signaling.so"

#### StateLess module
loadmodule "sl.so"

#### Transaction Module
loadmodule "tm.so"
modparam("tm", "fr_timeout", 5)
modparam("tm", "fr_inv_timeout", 30)
modparam("tm", "restart_fr_on_each_reply", 0)
modparam("tm", "onreply_avp_mode", 1)

#### Record Route Module
loadmodule "rr.so"
/* do not append from tag to the RR (no need for this script) */
modparam("rr", "append_fromtag", 0)

#### MAX ForWarD module
loadmodule "maxfwd.so"

#### SIP MSG OPerationS module
loadmodule "sipmsgops.so"

#### FIFO Management Interface
loadmodule "mi_fifo.so"
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("mi_fifo", "fifo_mode", 0666)

#### MYSQL module
loadmodule "db_mysql.so"

#### HTTPD module
loadmodule "httpd.so"
modparam("httpd", "port", 8888)

#### USeR LOCation module
loadmodule "usrloc.so"
modparam("usrloc", "nat_bflag", "NAT")
modparam("usrloc", "working_mode_preset", "single-instance-sql-write-back")
modparam("usrloc", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME


#### REGISTRAR module
loadmodule "registrar.so"
modparam("registrar", "tcp_persistent_flag", "TCP_PERSISTENT")
modparam("registrar", "received_avp", "$avp(received_nh)")/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)

#### ACCounting module
loadmodule "acc.so"
/* what special events should be accounted ? */
modparam("acc", "early_media", 0)
modparam("acc", "report_cancels", 0)
/* by default we do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure to enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
modparam("acc", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

#### AUTHentication modules
loadmodule "auth.so"
loadmodule "auth_db.so"
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME
modparam("auth_db", "load_credentials", "")

#### ALIAS module
loadmodule "alias_db.so"
modparam("alias_db", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

#### DIALOG module
loadmodule "dialog.so"
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 21600)  # 6 hours timeout
modparam("dialog", "db_mode", 2)
modparam("dialog", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

####  NAT modules
loadmodule "nathelper.so"
modparam("nathelper", "natping_interval", 10)
modparam("nathelper", "ping_nated_only", 1)
modparam("nathelper", "sipping_bflag", "SIP_PING_FLAG")
modparam("nathelper", "sipping_from", "sip:pinger at 127.0.0.1") #CUSTOMIZE ME
modparam("nathelper", "received_avp", "$avp(received_nh)")

loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:172.31.34.24:2225")

####  DIALPLAN module
loadmodule "dialplan.so"
modparam("dialplan", "db_url",
        "mysql://root:root at localhost/opensips") # CUSTOMIZE ME

####  MI_HTTP module
loadmodule "mi_http.so"

loadmodule "proto_udp.so"
loadmodule "proto_tcp.so"
loadmodule "proto_tls.so"
loadmodule "proto_ws.so"
loadmodule "proto_wss.so"
modparam("proto_wss", "wss_port", 7443)
modparam("proto_wss", "wss_max_msg_chunks", 16)

loadmodule "tls_openssl.so"
#loadmodule "tls_wolfssl.so"
loadmodule "tls_mgm.so"
modparam("tls_mgm","server_domain", "default")
modparam("tls_mgm","verify_cert", "[default]0")
modparam("tls_mgm","require_cert", "[default]0")
#modparam("tls_mgm","certificate", "[default]/etc/opensips/tls/rootCA/cacert.pem")
#modparam("tls_mgm","private_key", "[default]/etc/opensips/tls/rootCA/private/cakey.pem")
#modparam("tls_mgm","ca_list", "[bp3.erss.in]/etc/opensips/tls/user/user-calist.pem")
modparam("tls_mgm", "tls_method", "[default]SSLv23")
modparam("tls_mgm","tls_library","openssl")
#modparam("tls_mgm", "certificate", "[default]/etc/opensips/tls/user/caKey.pem")
#modparam("tls_mgm", "private_key", "[default]/etc/opensips/tls/user/privateKey.pem")
#modparam("tls_mgm", "client_domain", "14.139.
modparam("tls_mgm", "certificate", "[default]/home/ubuntu/cert.pem")
modparam("tls_mgm", "private_key", "[default]/home/ubuntu/privkey.pem")

####### Routing Logic ########

# main request routing logic
route{

        # initial NAT handling; detect if the request comes from behind a NAT
        # and apply contact fixing
        force_rport();
        if (nat_uac_test(23)) {
                if (is_method("REGISTER")) {
                        fix_nated_register();
                        setbflag("NAT");
                } else {
                        fix_nated_contact();
                        setflag("NAT");
                }
        }

        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;
                }

                # validate the sequential request against dialog
                if ( $DLG_status!=NULL && !validate_dialog() ) {
                        xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n");
                        ## exit;
                }

                if (is_method("BYE")) {
                        # do accounting even if the transaction fails
                        do_accounting("db","failed");

                }


                if (check_route_param("nat=yes"))
                        setflag("NAT");
                # 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")) {

                        # authenticate if from local subscriber
                        # authenticate all initial non-REGISTER request that pretend to be
                        # generated by local subscriber (domain from FROM URI is local)
                        if (!proxy_authorize("", "subscriber")) {
                                proxy_challenge("", "auth");
                                exit;
                        }
                        if ($au!=$fU) {
                                send_reply(403,"Forbidden auth ID");
                                exit;
                        }

                        consume_credentials();
                        # caller authenticated

                } 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("INVITE")) {
			# even if in most of the cases is useless, do RR for
			# re-INVITEs alos, as some buggy clients do change route set
			# during the dialog.
			record_route();
		}
                exit;
        }

        # record routing
        if (!is_method("REGISTER|MESSAGE"))
                record_route();

        # account only INVITEs
        if (is_method("INVITE")) {

                # create dialog with timeout
                if ( !create_dialog("B") ) {
                        send_reply(500,"Internal Server Error");
                        exit;
                }

                do_accounting("db");

        }


        if (!is_myself("$rd")) {
                append_hf("P-hint: outbound\r\n");

                # if you have some interdomain connections via TLS
                ## CUSTOMIZE IF NEEDED
                ##if ($rd=="tls_domain1.net"
                ## || $rd=="tls_domain2.net"
                ##) {
                ##      force_send_socket("tls:127.0.0.1:5061"); # CUSTOMIZE
                ##}

                route(relay);
        }
# requests for my domain

        if (is_method("PUBLISH|SUBSCRIBE")) {
                send_reply(503, "Service Unavailable");
                exit;
        }

        if (is_method("REGISTER")) {
                # authenticate the REGISTER requests
                if (!www_authorize("", "subscriber")) {
                        www_challenge("", "auth");
                        exit;
                }

                if ($au!=$tU) {
                        send_reply(403,"Forbidden auth ID");
                        exit;
                }
                if ($socket_in(proto) == "tcp" || $socket_in(proto) == "tls")
                        setflag("TCP_PERSISTENT");
                if (isflagset("NAT")) {
                        setbflag("SIP_PING_FLAG");
                }
                # store the registration and generate a SIP reply
                if (!save("location")){
                        sl_reply_error();
                        xlog("failed to register AoR $tu\n");
                        exit;
                }

                exit;
        }

        if ($rU==NULL) {
                # request with no Username in RURI
                send_reply(484,"Address Incomplete");
                exit;
        }
# apply DB based aliases
        alias_db_lookup("dbaliases");


        # apply transformations from dialplan table
        dp_translate( 0, "$rU", $rU);

        # check if the clients are using WebSockets or WebSocketSecure
        if ($socket_in(proto) == "WS" || $socket_in(proto) == "WSS")
                setflag('SRC_WS');
        else
                setflag('SRC_SIP');


        # consider the client is behind NAT - always fix the contact
        fix_nated_contact();

        if (is_method("REGISTER")) {

                # indicate that the client supports DTLS
                # so we know when he is called
                if (isflagset('SRC_WS'))
                        setbflag('DST_WS');

                fix_nated_register();
                if (!save("location"))
                        sl_reply_error();

                exit;
        }

        # do lookup with method filtering
        if (!lookup("location","m")) {
                if (!db_does_uri_exist("$ru","subscriber")) {
                        send_reply(420,"Bad Extension");
                        exit;
                }

                t_reply(404, "Not Found");
                exit;
        }
if (isbflagset("NAT")) setflag("NAT");


        # when routing via usrloc, log the missed calls also
        do_accounting("db","missed");

        route(relay);
}
route[relay] {
	# for INVITEs enable some additional helper routes
	if (is_method("INVITE")) {
		t_on_branch("handle_nat");
		t_on_reply("handle_nat");
	} else if (is_method("BYE|CANCEL")) {
		rtpengine_delete();
	}

	if (!t_relay()) {
		send_reply(500,"Internal Error");
	};
	exit;
}

branch_route[handle_nat] {

	if (!is_method("INVITE") || !has_body("application/sdp"))
		return;

	if (isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_offer("$var(rtpengine_flags)");
}

onreply_route[handle_nat] {

	fix_nated_contact();
	if (!has_body("application/sdp"))
		return;

	if (isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "ICE=force-relay DTLS=passive";
	else if (isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "UDP/TLS/RTP/SAVPF ICE=force";
	else if (!isflagset('SRC_WS') && isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";
	else if (!isflagset('SRC_WS') && !isbflagset('DST_WS'))
		$var(rtpengine_flags) = "RTP/AVP replace-session-connection replace-origin ICE=remove";

	rtpengine_answer("$var(rtpengine_flags)");

}

failure_route[missed_call] {
        if (t_was_cancelled()) {
                exit;
        }

        # uncomment the following lines if you want to block client 
        # redirect based on 3xx replies.
        ##if (t_check_status("3[0-9][0-9]")) {
        ##t_reply(404,"Not found");
        ##      exit;
        ##}


}

local_route {
        if (is_method("BYE") && $DLG_dir=="UPSTREAM") {

                acc_db_request("200 Dialog Timeout", "acc");

        }
}


From prathibhab.tvm at gmail.com  Tue Sep 26 08:11:19 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 26 Sep 2023 13:41:19 +0530
Subject: [OpenSIPS-Users] Temporarily unavailable
In-Reply-To: <CAJh_-cBfD40+uUnYoc7GD7EOZBuDSydbJEu8GkgBGiYp4cFj4w@mail.gmail.com>
References: <CAJh_-cBfD40+uUnYoc7GD7EOZBuDSydbJEu8GkgBGiYp4cFj4w@mail.gmail.com>
Message-ID: <CAJh_-cAQMxfn-mgeQjONXH1yB+8e76AcFxSQqWB5x26kuS2K=w@mail.gmail.com>

I am using websocket. I am trying web to web call.


On Tue, 26 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I've created two users and they are registered to opensips. When I try to
> connect, user 1 to user 2, I get temporarily unavailable. Config file is
> attached with this email.
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From bogdan at opensips.org  Tue Sep 26 08:19:44 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 26 Sep 2023 11:19:44 +0300
Subject: [OpenSIPS-Users] Wrong TCP socket being used on TLS
	registrations
In-Reply-To: <d1ee08b6-5545-1dbb-6423-a8241a07d100@hero.co.nz>
References: <864e2852-950a-9cff-f38e-9910e344b417@hero.co.nz>
 <1e279a15-ff97-019a-3e62-55d7eab5009b@opensips.org>
 <d1ee08b6-5545-1dbb-6423-a8241a07d100@hero.co.nz>
Message-ID: <8ab42281-7c55-1480-7e62-0440e985f0d2@opensips.org>

Hi Ray,

The "tcp_accept_aliases" should be harmless if there is no "alias" param 
received in the incoming requests. If no such parameter is pushed by the 
end-devices, there is 0 impact.

And indeed, this has a really ugly side effect for the (CG)NAT'd 
devices. But let's give it a try, disable this option and try the 
testing again, to see if the right conn is selected.

Regards,

Bogdan-Andrei Iancu

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

On 9/22/23 5:03 AM, Ray Jackson wrote:
>
> Hi Bogdan,
>
> Yes, we have the following enabled in our config:
>
> tcp_accept_aliases=1
>
> I assume this is the culprit then and we are inadvertently sending 
> calls down the wrong TCP socket here to the wrong user due to this 
> being enabled?  This is quite a nasty setting to have enabled when we 
> are dealing with CGNAT'd customers who are sharing public IP addresses 
> but are completely unrelated users!
>
> I will disable this setting and see if that clears up the issue for 
> us.  We have in fact had another case just today of the same issue 
> happening (User A is receiving User B's incoming calls!)
>
> Thanks for highlighting this and let me know if there is anything else 
> I should look at in our config.
>
> Thanks,
>
> Ray
>
> On 19/09/23 9:30 pm, Bogdan-Andrei Iancu wrote:
>
>> Hi Ray,
>>
>> Do you use any TCP aliasing options in your cfg ?
>>
>> Regards,
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>    https://www.opensips-solutions.com
>>    https://www.siphub.com
>> On 9/2/23 3:17 AM, Ray Jackson wrote:
>>>
>>> Hi all,
>>>
>>> I'm facing a weird issue which I think is related to broken TCP 
>>> socket reuse logic where the wrong client is receiving incoming 
>>> calls due to the wrong socket being used for the incoming INVITE.
>>>
>>> The scenario is when I have 2 clients registering using TLS behind 
>>> NAT at the same Public IPv4 address and both clients are using the 
>>> same private port number.  So client 1 registers and the Via and 
>>> contact header looks like:
>>>
>>> Via: SIP/2.0/TLS 
>>> 192.168.42.162:5062;branch=z9hG4bK1409895926;rport;alias Contact: 
>>> <sip:201 at 192.168.42.162:5062;transport=tls>;reg-id=2;+sip.instance="<urn:uuid:00000000-0000-1000-8000-C074AD928AC4>"
>>>
>>> Client 2 registers from behind the same Public IPv4 address and the 
>>> Via and contact header looks like:
>>>
>>> Via: SIP/2.0/TLS 192.168.42.186:5062;branch=z9hG4bK-aff1f3b3 
>>> Contact: <sip:202 at 192.168.42.186:5062;transport=tls>;expires=300
>>>
>>> The location table shows Client 1 received field of 103.212.1.2:5062 
>>> and Client 103.212.1.2:23456
>>>
>>> When a call comes in for Client 1 the location lookup seems to 
>>> return the correct 'received' address and port (e.g. 
>>> 103.212.1.2:5062) and all the logs indicate that this is where the 
>>> SIP INVITE *should* be going to (in the $du field).  However when 
>>> you check the SIP traffic it selects Client 2's socket and the 
>>> traffic goes to port 23456 instead of 5062.
>>>
>>> I think this is related somehow to the TCP port reuse logic inside 
>>> Opensips.  My suspicion is that Opensips is looking at the Contact 
>>> or Via port number (which is the same for both client 1 and 2) and 
>>> then somehow mapping this to the wrong TCP received socket.
>>>
>>> Does anybody have any suggestions here?  Should I be fixing the NAT 
>>> in the Contact header (using fix_nated_contact).  I read somewhere 
>>> that you shouldn't rewrite the Contact header to avoid problems with 
>>> sending a different Contact URI to the client on calls.  Or is this 
>>> issue more related to the Via header and the TCP port reuse logic 
>>> looking at this port instead of the actual received port when 
>>> choosing the outgoing socket?
>>>
>>> FYI: I am using both force_rport() and fix_nated_register() for 
>>> incoming registrations from these clients and matching_mode of 0 in 
>>> usrloc.  However, I am not using fix_nated_contact() for registrations.
>>>
>>> Thanks,
>>>
>>> Ray
>>>
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>

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From prathibhab.tvm at gmail.com  Tue Sep 26 08:38:53 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 26 Sep 2023 14:08:53 +0530
Subject: [OpenSIPS-Users] Temporarily unavailable
In-Reply-To: <CAJh_-cAQMxfn-mgeQjONXH1yB+8e76AcFxSQqWB5x26kuS2K=w@mail.gmail.com>
References: <CAJh_-cBfD40+uUnYoc7GD7EOZBuDSydbJEu8GkgBGiYp4cFj4w@mail.gmail.com>
 <CAJh_-cAQMxfn-mgeQjONXH1yB+8e76AcFxSQqWB5x26kuS2K=w@mail.gmail.com>
Message-ID: <CAJh_-cD9BqFHJafvdQv3fbLgTpJrEqYG99JhyvmBe+c+gS5sNQ@mail.gmail.com>

I'm getting this error in log:

 New TLS connection from 45.33.68.226:42316 failed to accept
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A0000EA:SSL
routines::callback failed
ERROR:proto_wss:wss_read_req: cannot fix read connection

On Tue, 26 Sept 2023 at 13:41, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I am using websocket. I am trying web to web call.
>
>
> On Tue, 26 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> I've created two users and they are registered to opensips. When I try to
>> connect, user 1 to user 2, I get temporarily unavailable. Config file is
>> attached with this email.
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From prathibhab.tvm at gmail.com  Tue Sep 26 08:46:07 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 26 Sep 2023 14:16:07 +0530
Subject: [OpenSIPS-Users] Temporarily unavailable
In-Reply-To: <CAJh_-cD9BqFHJafvdQv3fbLgTpJrEqYG99JhyvmBe+c+gS5sNQ@mail.gmail.com>
References: <CAJh_-cBfD40+uUnYoc7GD7EOZBuDSydbJEu8GkgBGiYp4cFj4w@mail.gmail.com>
 <CAJh_-cAQMxfn-mgeQjONXH1yB+8e76AcFxSQqWB5x26kuS2K=w@mail.gmail.com>
 <CAJh_-cD9BqFHJafvdQv3fbLgTpJrEqYG99JhyvmBe+c+gS5sNQ@mail.gmail.com>
Message-ID: <CAJh_-cAwtz9NL2704LaQp3s242gtcu+QcNNpB93wP8F+foiOGg@mail.gmail.com>

When user2 accepts the call, the above error occurs.

On Tue, 26 Sept 2023 at 14:08, Prathibha B <prathibhab.tvm at gmail.com> wrote:

> I'm getting this error in log:
>
>  New TLS connection from 45.33.68.226:42316 failed to accept
> ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A0000EA:SSL
> routines::callback failed
> ERROR:proto_wss:wss_read_req: cannot fix read connection
>
> On Tue, 26 Sept 2023 at 13:41, Prathibha B <prathibhab.tvm at gmail.com>
> wrote:
>
>> I am using websocket. I am trying web to web call.
>>
>>
>> On Tue, 26 Sept 2023 at 13:40, Prathibha B <prathibhab.tvm at gmail.com>
>> wrote:
>>
>>> I've created two users and they are registered to opensips. When I try
>>> to connect, user 1 to user 2, I get temporarily unavailable. Config file is
>>> attached with this email.
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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From vincent.horst at ehvc.nl  Tue Sep 26 10:27:56 2023
From: vincent.horst at ehvc.nl (Vincent Horst)
Date: Tue, 26 Sep 2023 12:27:56 +0200
Subject: [OpenSIPS-Users] Opensips 3.4 Not sending hep to homer server
In-Reply-To: <CAPWMwMF1AbxfWJ8cx8rcuH=S4NWTauaCCvDXT3RB73Wow8XYRQ@mail.gmail.com>
References: <CAK1AzRXtMTe8z-R6GxRkO1PsmCYa8Aqq=0x1otrwwxg-xtx2gw@mail.gmail.com>
 <CAPWMwMF1AbxfWJ8cx8rcuH=S4NWTauaCCvDXT3RB73Wow8XYRQ@mail.gmail.com>
Message-ID: <CAK1AzRWtx6FsBnBHaEiRFhe6drBUeFXTG9YxazRV2X57=RT7Gg@mail.gmail.com>

Thanks @Michel,

hep socket is indeed the local host.
step I missed, installing heplify agent on the opensips system.
Forward the received packets towards Homer :)

case closed.
Vincent


Op di 26 sep 2023 om 08:21 schreef Michel crans <mcrans at gmail.com>:

> If you're not seeing any packets received on the Homer server, there could
> be several reasons for this issue. Let's go through some troubleshooting
> steps:
>
>    1.
>
>    Check Network Connectivity:
>    - Ensure that there is network connectivity between your OpenSIPS
>       server and the Homer server. You should be able to ping the Homer server
>       from the OpenSIPS server.
>    2.
>
>    Homer Server Configuration:
>    - Verify that your Homer server is correctly configured to receive HEP
>       packets on the specified IP and port (in this case, 127.0.0.1:6061).
>       Check Homer's logs for any incoming connection attempts or errors.
>    3.
>
>    Socket Configuration:
>    - It seems you are using 127.0.0.1 as the IP address in your socket
>       configuration (socket=hep_udp:127.0.0.1:6061). This means OpenSIPS
>       will send HEP packets to localhost (the same server it's running on).
>       Ensure that this is the correct configuration. If Homer is on a different
>       server, you should use the Homer server's IP address instead.
>    4.
>
>    Firewall and Security Rules:
>    - Make sure that there are no firewall rules or security policies
>       blocking traffic on the specified HEP port (6061) or the network interface
>       being used for communication.
>    5.
>
>    Logging and Debugging:
>    - Enable verbose logging in OpenSIPS to see if there are any error
>       messages related to HEP. You can do this by setting the log_level
>       parameter in your OpenSIPS configuration file.
>
>    cfgCopy code
>    modparam("log", "log_level", 3)
>
>    This will increase the logging verbosity, and you can check the
>    OpenSIPS logs for any relevant error messages.
>    6.
>
>    Verify Route Execution:
>    - Ensure that your route[to_homer] block is being executed when you
>       expect it to. You can add additional log messages in this block to verify
>       that it's being triggered.
>    7.
>
>    Check HEP Configuration:
>    - Review your HEP configuration parameters, such as hep_id and trace_id,
>       to ensure they are correctly set. Make sure they match the configuration on
>       the Homer server.
>    8.
>
>    Firewall on Homer Server:
>    - Check if there is any firewall or security software running on the
>       Homer server that might be blocking incoming HEP packets.
>    9.
>
>    Homer Server Logs:
>    - Examine the logs on your Homer server to see if there are any error
>       messages or indications of failed connections.
>    10.
>
>    Packet Capture:
>    - You can use tools like Wireshark to capture network traffic on the
>       OpenSIPS server and check if the HEP packets are actually being sent. This
>       can help you confirm whether the issue is with OpenSIPS or the network.
>
> By systematically going through these troubleshooting steps, you should be
> able to identify the root cause of why OpenSIPS is not sending HEP packets
> to your Homer server. Remember to make configuration adjustments as needed
> to match your specific network setup and requirements.
>
> Op ma 25 sep 2023 om 22:34 schreef Vincent Horst via Users <
> users at lists.opensips.org>:
>
>> Hi All,
>>
>> I have a issue where opensips is not sending the hep files towards homer.
>> I have used a manual and have add below to the .cfg file.
>>
>> Sockets: (It is unclear for my why I should put here the local adres,
>> when I put the remote IP of homer, I get an error in the config:
>> socket=hep_udp:127.0.0.1:6061
>> socket=hep_tcp:127.0.0.1:6061
>> If I remove them from the config the hep module gives an error.
>>
>> Second part in the .cfg:
>> ### Configure an HEP Endpoint
>> loadmodule "proto_hep.so"
>> modparam("proto_hep", "hep_id", "[hid]10.0.0.216:9060
>> ;transport=udp;version=3")
>> #### Configure Tracer module to use the HEP Protocol instance id [hid]
>> loadmodule "tracer.so"
>> modparam("tracer", "trace_on", 1)
>> modparam("tracer", "trace_id", "[tid]uri=hep:hid")
>>
>> And then the third part,
>> I noticed that it make no different when I put it above in the route
>> logic or below.
>>
>> route[to_homer] {
>> $var(trace_id) = "tid";
>> if (!has_totag()) {
>> if (is_method("INVITE")) { $var(trace_type) = "dialog"; }
>> else if (!is_method("CANCEL")) { $var(trace_type) = "transaction"; }
>> } else { $var(trace_type) = NULL; }
>> switch ($var(trace_type)) {
>> case "dialog":
>> trace("$var(trace_id)", "d", "sip|xlog|rest");
>> break;
>> case "transaction":
>> trace("$var(trace_id)", "t", "sip|xlog");
>> break;
>> }
>> }
>>
>> I'm not getting an error, but I'm also not seeing any packets received on
>> the homer server.
>> Can someone point me in the right direction or explain how I can add
>> details to trouble shoot?
>>
>> Cheers,
>> Vincent
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>

-- 
Met vriendelijke groet,

Vincent Horst
EHVC Zakelijk
www.ehvc.nl
VincentHorst at EHVC.nl
Tel: 06 518 218 48

DISCLAIMER

Dit bericht is uitsluitend bestemd voor de geadresseerde. Als u dit bericht
per abuis heeft ontvangen, wordt u verzocht het te vernietigen en de
afzender te informeren. EHVC Zakelijk wijst elke aansprakelijkheid af die
voortvloeit uit elektronische verzending.
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From prathibhab.tvm at gmail.com  Tue Sep 26 11:47:47 2023
From: prathibhab.tvm at gmail.com (Prathibha B)
Date: Tue, 26 Sep 2023 17:17:47 +0530
Subject: [OpenSIPS-Users] unexpect EOF while reading
Message-ID: <CAJh_-cCd4tE5+KbxJcipXQ7SRqCROd08OLMr2=3GVy7BH+KOdw@mail.gmail.com>

opensips version: 3.3

Error:
I am using websocket. I am using the letsencrypt certifcate. When I
connected from web to web and when verify_cert is 1 and require_cert is 1,
get the following errors:

ERROR:tls_openssl:openssl_tls_accept: New TLS connection from
14.139.183.221:53946 failed to accept
ERROR:tls_openssl:tls_print_errstack: TLS errstack: error:0A000126:SSL
routines::unexpected eof while reading
ERROR:proto_wss:wss_read_req: cannot fix read connection

-- 
Regards,
B.Prathibha
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From bogdan at opensips.org  Tue Sep 26 12:46:06 2023
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 26 Sep 2023 15:46:06 +0300
Subject: [OpenSIPS-Users] Code of conduct
Message-ID: <d9958782-f624-2382-ace7-a237ccc0e737@opensips.org>

Hi,

Following the abundance of emails and GH tickets from B.Prathibha, I 
want to point out some important aspects of the code of conduct here:

1) BEFORE considering the community help, be sure you checked the 
available documentation and did any online research on the matter

2) BEFORE consider posting, pick the right channel - if you suspect a 
coding bug, use the GitHub ticket, otherwise post on the user's mailing list

3) DO NOT DOUBLE POST on several channels (list and GH tracker)

4) POST only when you understand the problem you have - avoid fast 
rounds of posts where you are just scratchpad'ing

5) POST only for topics related to OpenSIPS / SIP

6) DO NOT ASK people to solve you problem or do your script - ask for 
guidance, help and info


Keep in mind, the community help is here for all, so respect the rest of 
the people on the list and AVOID spamming or abusing the channels we have.

Best regards,

-- 
Bogdan-Andrei Iancu

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



From efes99999 at gmail.com  Tue Sep 26 13:38:51 2023
From: efes99999 at gmail.com (L S)
Date: Tue, 26 Sep 2023 09:38:51 -0400
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
Message-ID: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>

I'm trying to create certificates using opensips-cli:

opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
DEBUG: Skipping module 'tls' - excluded on purpose

ERROR: No module 'tls' loaded

Trying to find out why I am getting this message now - it used to work
fine. All other modules are loaded.

Thaks,
Matt
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From razvan at opensips.org  Tue Sep 26 13:52:03 2023
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Tue, 26 Sep 2023 16:52:03 +0300
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
In-Reply-To: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
References: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
Message-ID: <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>

Can you double check whether you have the python-openssl or 
python-cryptography libraries?

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/26/23 16:38, L S wrote:
> I'm trying to create certificates using opensips-cli:
> 
> opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
> DEBUG: Skipping module 'tls' - excluded on purpose
> 
> ERROR: No module 'tls' loaded
> 
> Trying to find out why I am getting this message now - it used to work 
> fine. All other modules are loaded.
> 
> Thaks,
> Matt
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From efes99999 at gmail.com  Tue Sep 26 16:54:20 2023
From: efes99999 at gmail.com (L S)
Date: Tue, 26 Sep 2023 12:54:20 -0400
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
In-Reply-To: <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
References: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
 <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
Message-ID: <CAPVhdZ9duJknc7cYpEpnai1eGi8Thc4FWy8cHfWtw+072ByfTA@mail.gmail.com>

Thanks Razvan. Installing the cryptography module fixed it - I was able to
run both -x tls rootCA and userCERT, and create the certificates.

However, when I start Opensips, I get the following error:
ERROR:tls_wolfssl:load_private_key: key
'/usr/local/etc/opensips/tls/server/privkey.pem' does not match the public
key of the certificate

I tried creating the certificates both on Centos 7 and Ubuntu Focal, and
they both gave the same error.
The data for the certificates comes from opensips-cli.cfg. I had created
certificates with that cfg 3 months ago, and used in Opensips script
without any issues.
 I only changed the domain name this time.

Any suggestions?
Thanks,
Matt


On Tue, Sep 26, 2023, 9:56 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Can you double check whether you have the python-openssl or
> python-cryptography libraries?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
>
> On 9/26/23 16:38, L S wrote:
> > I'm trying to create certificates using opensips-cli:
> >
> > opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
> > DEBUG: Skipping module 'tls' - excluded on purpose
> >
> > ERROR: No module 'tls' loaded
> >
> > Trying to find out why I am getting this message now - it used to work
> > fine. All other modules are loaded.
> >
> > Thaks,
> > Matt
> >
> > _______________________________________________
> > 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 efes99999 at gmail.com  Tue Sep 26 18:15:58 2023
From: efes99999 at gmail.com (L S)
Date: Tue, 26 Sep 2023 14:15:58 -0400
Subject: [OpenSIPS-Users] Multiple TLS server domain setup
Message-ID: <CAPVhdZ-5WYBYij96kdmqXc9RE6O9cO_v_X+pijwksggJCS8prg@mail.gmail.com>

Hi,
I'm trying to set up two tls domains for two sets of clients. First one
requires TLSv1 (higher not supported), and the other one requires TLSv1_2
or higher.
Right now the domain with tlsv1 is active on 5061 and has no issues. I'm
trying to add the second domain.

As far as I understand (do not have much experience with tls config), for
incoming traffic (server domain), we can either ask them to use port 5062
or provide SNI so that they can also connect thru 5061. Not sure if they
want to/can do that. Is there any other way we can distinguish these two
clients; e.g. from the source ip?

Thanks,
Matt
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From John.Sliney at lcs.com  Tue Sep 26 18:28:34 2023
From: John.Sliney at lcs.com (John Sliney)
Date: Tue, 26 Sep 2023 18:28:34 +0000
Subject: [OpenSIPS-Users] dbalias and location lookup branching
Message-ID: <CO6P222MB0570536DEEBE3DC963F05FABE0C3A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>

Hi,

I’m currently attempting to take an INVITE from an Asterisk server that is requesting an extension number, perform dbalias lookups to have extensions turned into sip users (x1000 -> test_hardphone) and then do location lookups on those sip users.  There can be multiple sip users for each extension and multiple locations for each sip user.

Using the code below partially works but when there are no location entries for the requested sip user, OpenSIPS returns a 500 and prints out “ERROR:  t_forward_nonack failed” how can I have OpenSIPS instead respond with a 404?

### CODE ###

modparam("alias_db",        "append_branches",          1)

route {
    if ( is_from_gw() ) {
        alias_db_lookup("dbaliases");
        t_on_branch("sip_user_branch");
    }
    route(relay);
    return(0);
}

branch_route[sip_user_branch] {
    route(lookup_sip_user);
}

route[lookup_sip_user] {
    if ( ! lookup("location")  ) {
        drop();
    }
}

route[relay] {
    if ( ! t_relay() ) {
        sl_reply_error();
        rtpproxy_unforce();
        return(0);
    }
}

### CODE ###

Any help would be appreciated, Thanks
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From efes99999 at gmail.com  Tue Sep 26 18:42:51 2023
From: efes99999 at gmail.com (L S)
Date: Tue, 26 Sep 2023 14:42:51 -0400
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
In-Reply-To: <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
References: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
 <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
Message-ID: <CAPVhdZ-yQW1+m=68=GxBFbmQG+RMmYK_+6XxmDBEraaswrmMyw@mail.gmail.com>

I apologize if this is a duplicate post - ran into some errors while
posting the first time.

Thanks Razvan. Installing cryptography fixed that issue. I was able to run
-x tls rootCA and userCERT, and create the certificates.

However, when running Opensips I get this error now:

ERROR:tls_wolfssl:load_private_key: key
'/usr/local/etc/opensips/tls/server/privkey.pem' does not match the public
key of the certificate

I had created and used certificates with opensips-cli before without any
issues. Opensips-cli.cfg is the same except for a small change to CN. All
the paths are the same as before and correct.

I compared the modulus of the server private key to the public key using
openssl, and they are different. Btw I created certificates both in Centos
7 and Ubuntu Focal just to see if it matters; got the same error for both.

Any ideas why this is happening?

Thanks,
Matt

On Tue, Sep 26, 2023, 9:56 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Can you double check whether you have the python-openssl or
> python-cryptography libraries?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
>
> On 9/26/23 16:38, L S wrote:
> > I'm trying to create certificates using opensips-cli:
> >
> > opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
> > DEBUG: Skipping module 'tls' - excluded on purpose
> >
> > ERROR: No module 'tls' loaded
> >
> > Trying to find out why I am getting this message now - it used to work
> > fine. All other modules are loaded.
> >
> > Thaks,
> > Matt
> >
> > _______________________________________________
> > 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 medeanwz at gmail.com  Wed Sep 27 10:54:05 2023
From: medeanwz at gmail.com (M S)
Date: Wed, 27 Sep 2023 12:54:05 +0200
Subject: [OpenSIPS-Users] dbalias and location lookup branching
In-Reply-To: <CO6P222MB0570536DEEBE3DC963F05FABE0C3A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>
References: <CO6P222MB0570536DEEBE3DC963F05FABE0C3A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>
Message-ID: <CAEZrhoNaRfRi7L1ZGnfem+C91LTx-qPTM4pXH9Js+MLXAFcsqQ@mail.gmail.com>

Maybe a send_reply(404) after if(!lookup(location)), instead of drop?

On Tue, Sep 26, 2023 at 8:30 PM John Sliney <John.Sliney at lcs.com> wrote:

> Hi,
>
> I’m currently attempting to take an INVITE from an Asterisk server that is
> requesting an extension number, perform dbalias lookups to have extensions
> turned into sip users (x1000 -> test_hardphone) and then do location
> lookups on those sip users.  There can be multiple sip users for each
> extension and multiple locations for each sip user.
>
> Using the code below partially works but when there are no location
> entries for the requested sip user, OpenSIPS returns a 500 and prints out
> “ERROR:  t_forward_nonack failed” how can I have OpenSIPS instead respond
> with a 404?
>
> ### CODE ###
>
> modparam("alias_db",        "append_branches",          1)
>
> route {
>     if ( is_from_gw() ) {
>         alias_db_lookup("dbaliases");
>         t_on_branch("sip_user_branch");
>     }
>     route(relay);
>     return(0);
> }
>
> branch_route[sip_user_branch] {
>     route(lookup_sip_user);
> }
>
> route[lookup_sip_user] {
>     if ( ! lookup("location")  ) {
>         drop();
>     }
> }
>
> route[relay] {
>     if ( ! t_relay() ) {
>         sl_reply_error();
>         rtpproxy_unforce();
>         return(0);
>     }
> }
>
> ### CODE ###
>
> Any help would be appreciated, Thanks
> _______________________________________________
> 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  Wed Sep 27 13:47:44 2023
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Wed, 27 Sep 2023 16:47:44 +0300
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
In-Reply-To: <CAPVhdZ9duJknc7cYpEpnai1eGi8Thc4FWy8cHfWtw+072ByfTA@mail.gmail.com>
References: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
 <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
 <CAPVhdZ9duJknc7cYpEpnai1eGi8Thc4FWy8cHfWtw+072ByfTA@mail.gmail.com>
Message-ID: <7d18f328-721e-4e80-99f5-b5cc84b0e1a2@opensips.org>

Can you actually check that the two (private key and certificate) match?
https://www.ibm.com/support/pages/how-verify-if-private-key-matches-certificate

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/26/23 19:54, L S wrote:
> Thanks Razvan. Installing the cryptography module fixed it - I was able 
> to run both -x tls rootCA and userCERT, and create the certificates.
> 
> However, when I start Opensips, I get the following error:
> ERROR:tls_wolfssl:load_private_key: key 
> '/usr/local/etc/opensips/tls/server/privkey.pem' does not match the 
> public key of the certificate
> 
> I tried creating the certificates both on Centos 7 and Ubuntu Focal, and 
> they both gave the same error.
> The data for the certificates comes from opensips-cli.cfg. I had created 
> certificates with that cfg 3 months ago, and used in Opensips script 
> without any issues.
>   I only changed the domain name this time.
> 
> Any suggestions?
> Thanks,
> Matt
> 
> 
> On Tue, Sep 26, 2023, 9:56 AM Răzvan Crainea <razvan at opensips.org 
> <mailto:razvan at opensips.org>> wrote:
> 
>     Can you double check whether you have the python-openssl or
>     python-cryptography libraries?
> 
>     Best regards,
> 
>     Răzvan Crainea
>     OpenSIPS Core Developer / SIPhub CTO
>     http://www.opensips-solutions.com
>     <http://www.opensips-solutions.com> / https://www.siphub.com
>     <https://www.siphub.com>
> 
>     On 9/26/23 16:38, L S wrote:
>      > I'm trying to create certificates using opensips-cli:
>      >
>      > opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
>      > DEBUG: Skipping module 'tls' - excluded on purpose
>      >
>      > ERROR: No module 'tls' loaded
>      >
>      > Trying to find out why I am getting this message now - it used to
>     work
>      > fine. All other modules are loaded.
>      >
>      > Thaks,
>      > Matt
>      >
>      > _______________________________________________
>      > 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


From razvan at opensips.org  Wed Sep 27 13:52:20 2023
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Wed, 27 Sep 2023 16:52:20 +0300
Subject: [OpenSIPS-Users] Multiple TLS server domain setup
In-Reply-To: <CAPVhdZ-5WYBYij96kdmqXc9RE6O9cO_v_X+pijwksggJCS8prg@mail.gmail.com>
References: <CAPVhdZ-5WYBYij96kdmqXc9RE6O9cO_v_X+pijwksggJCS8prg@mail.gmail.com>
Message-ID: <ce7244f6-d291-4ae1-b7b3-881a03a96017@opensips.org>

Unfortunately no, it's either SNI, or a different port. There's 
currently no way to filter based on source IP address.

Best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/26/23 21:15, L S wrote:
> Hi,
> I'm trying to set up two tls domains for two sets of clients. First one 
> requires TLSv1 (higher not supported), and the other one requires 
> TLSv1_2 or higher.
> Right now the domain with tlsv1 is active on 5061 and has no issues. I'm 
> trying to add the second domain.
> 
> As far as I understand (do not have much experience with tls config), 
> for incoming traffic (server domain), we can either ask them to use port 
> 5062 or provide SNI so that they can also connect thru 5061. Not sure if 
> they want to/can do that. Is there any other way we can distinguish 
> these two clients; e.g. from the source ip?
> 
> Thanks,
> Matt
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From razvan at opensips.org  Wed Sep 27 14:00:19 2023
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Wed, 27 Sep 2023 17:00:19 +0300
Subject: [OpenSIPS-Users] dbalias and location lookup branching
In-Reply-To: <CAEZrhoNaRfRi7L1ZGnfem+C91LTx-qPTM4pXH9Js+MLXAFcsqQ@mail.gmail.com>
References: <CO6P222MB0570536DEEBE3DC963F05FABE0C3A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>
 <CAEZrhoNaRfRi7L1ZGnfem+C91LTx-qPTM4pXH9Js+MLXAFcsqQ@mail.gmail.com>
Message-ID: <92e86a5c-1d8e-4c06-af72-fd7165be971b@opensips.org>

No, this is not the solution :).

The problem is you are calling t_relay, but after you evaluate each 
branch, it turns out t_relay does not actually relay anything, hence it 
returns an error.

The proper way to do this is to figure out whether you do need to send 
any branches before calling t_relay() - this means that after the 
alias_db_lookup, you can simply call the lookup() function - if that 
fails, you should reply with a 404.
But what I am missing is the 500 message - what happened with the call 
to the actual extension? Because from your script, it appears you still 
want to keep it as a branch, don't you? Isn't that branch properly sent?

Also, the 500 sent to the client is very likely sent by the 
sl_reply_error() function - you can replace it with a 404. However, I'd 
still refactor everything to handle the branches in the main processing 
context, not in the branch route.

My 2cents, best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/27/23 13:54, M S wrote:
> Maybe a send_reply(404) after if(!lookup(location)), instead of drop?
> 
> On Tue, Sep 26, 2023 at 8:30 PM John Sliney <John.Sliney at lcs.com 
> <mailto:John.Sliney at lcs.com>> wrote:
> 
>     Hi,
> 
>     I’m currently attempting to take an INVITE from an Asterisk server
>     that is requesting an extension number, perform dbalias lookups to
>     have extensions turned into sip users (x1000 -> test_hardphone) and
>     then do location lookups on those sip users.  There can be multiple
>     sip users for each extension and multiple locations for each sip user.
> 
>     Using the code below partially works but when there are no location
>     entries for the requested sip user, OpenSIPS returns a 500 and
>     prints out “ERROR:  t_forward_nonack failed” how can I
>     have OpenSIPS instead respond with a 404?
> 
>     ### CODE ###
> 
>     modparam("alias_db",        "append_branches",          1)
> 
>     route {
>          if ( is_from_gw() ) {
>              alias_db_lookup("dbaliases");
>              t_on_branch("sip_user_branch");
>          }
>          route(relay);
>          return(0);
>     }
> 
>     branch_route[sip_user_branch] {
>          route(lookup_sip_user);
>     }
> 
>     route[lookup_sip_user] {
>          if ( ! lookup("location")  ) {
>              drop();
>          }
>     }
> 
>     route[relay] {
>          if ( ! t_relay() ) {
>              sl_reply_error();
>              rtpproxy_unforce();
>              return(0);
>          }
>     }
> 
>     ### CODE ###
> 
>     Any help would be appreciated, Thanks
>     _______________________________________________
>     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


From efes99999 at gmail.com  Wed Sep 27 15:55:58 2023
From: efes99999 at gmail.com (L S)
Date: Wed, 27 Sep 2023 11:55:58 -0400
Subject: [OpenSIPS-Users] opensips-cli skipping module tls
In-Reply-To: <7d18f328-721e-4e80-99f5-b5cc84b0e1a2@opensips.org>
References: <CAPVhdZ82_Grv6++d1Ctdsm4wq4_AuhykaLBdB+y45XjnqZ7H5g@mail.gmail.com>
 <a20ce549-3067-4bf4-b4b7-7ef6d58b3b61@opensips.org>
 <CAPVhdZ9duJknc7cYpEpnai1eGi8Thc4FWy8cHfWtw+072ByfTA@mail.gmail.com>
 <7d18f328-721e-4e80-99f5-b5cc84b0e1a2@opensips.org>
Message-ID: <CAPVhdZ8DhtbZrGLjLmE6wDD+BuG64RC7KJ-a=-htkbzoR9xCEA@mail.gmail.com>

Hi Razvan,
They don't match. Not sure if sth on my end causing this problem. I was
using opensips-cli only to create the certificates.

Anyway I used openssl directly instead to create the CA and server
certificates. They are working fine.

Thanks,
Matt

On Wed, Sep 27, 2023, 9:50 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Can you actually check that the two (private key and certificate) match?
>
> https://www.ibm.com/support/pages/how-verify-if-private-key-matches-certificate
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
>
> On 9/26/23 19:54, L S wrote:
> > Thanks Razvan. Installing the cryptography module fixed it - I was able
> > to run both -x tls rootCA and userCERT, and create the certificates.
> >
> > However, when I start Opensips, I get the following error:
> > ERROR:tls_wolfssl:load_private_key: key
> > '/usr/local/etc/opensips/tls/server/privkey.pem' does not match the
> > public key of the certificate
> >
> > I tried creating the certificates both on Centos 7 and Ubuntu Focal, and
> > they both gave the same error.
> > The data for the certificates comes from opensips-cli.cfg. I had created
> > certificates with that cfg 3 months ago, and used in Opensips script
> > without any issues.
> >   I only changed the domain name this time.
> >
> > Any suggestions?
> > Thanks,
> > Matt
> >
> >
> > On Tue, Sep 26, 2023, 9:56 AM Răzvan Crainea <razvan at opensips.org
> > <mailto:razvan at opensips.org>> wrote:
> >
> >     Can you double check whether you have the python-openssl or
> >     python-cryptography libraries?
> >
> >     Best regards,
> >
> >     Răzvan Crainea
> >     OpenSIPS Core Developer / SIPhub CTO
> >     http://www.opensips-solutions.com
> >     <http://www.opensips-solutions.com> / https://www.siphub.com
> >     <https://www.siphub.com>
> >
> >     On 9/26/23 16:38, L S wrote:
> >      > I'm trying to create certificates using opensips-cli:
> >      >
> >      > opensips-cli - f /usr/local/etc/opensips-cli.cfg -d -x tls rootCA
> >      > DEBUG: Skipping module 'tls' - excluded on purpose
> >      >
> >      > ERROR: No module 'tls' loaded
> >      >
> >      > Trying to find out why I am getting this message now - it used to
> >     work
> >      > fine. All other modules are loaded.
> >      >
> >      > Thaks,
> >      > Matt
> >      >
> >      > _______________________________________________
> >      > 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
>
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From efes99999 at gmail.com  Wed Sep 27 15:56:26 2023
From: efes99999 at gmail.com (L S)
Date: Wed, 27 Sep 2023 11:56:26 -0400
Subject: [OpenSIPS-Users] Multiple TLS server domain setup
In-Reply-To: <ce7244f6-d291-4ae1-b7b3-881a03a96017@opensips.org>
References: <CAPVhdZ-5WYBYij96kdmqXc9RE6O9cO_v_X+pijwksggJCS8prg@mail.gmail.com>
 <ce7244f6-d291-4ae1-b7b3-881a03a96017@opensips.org>
Message-ID: <CAPVhdZ9Tdb7qwg7hrFXQAvjCZACcAuYHBF+=xiCW0VxZgQpdag@mail.gmail.com>

Thanks Razvan.

On Wed, Sep 27, 2023, 9:55 AM Răzvan Crainea <razvan at opensips.org> wrote:

> Unfortunately no, it's either SNI, or a different port. There's
> currently no way to filter based on source IP address.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com / https://www.siphub.com
>
> On 9/26/23 21:15, L S wrote:
> > Hi,
> > I'm trying to set up two tls domains for two sets of clients. First one
> > requires TLSv1 (higher not supported), and the other one requires
> > TLSv1_2 or higher.
> > Right now the domain with tlsv1 is active on 5061 and has no issues. I'm
> > trying to add the second domain.
> >
> > As far as I understand (do not have much experience with tls config),
> > for incoming traffic (server domain), we can either ask them to use port
> > 5062 or provide SNI so that they can also connect thru 5061. Not sure if
> > they want to/can do that. Is there any other way we can distinguish
> > these two clients; e.g. from the source ip?
> >
> > Thanks,
> > Matt
> >
> > _______________________________________________
> > 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 John.Sliney at lcs.com  Wed Sep 27 16:01:24 2023
From: John.Sliney at lcs.com (John Sliney)
Date: Wed, 27 Sep 2023 16:01:24 +0000
Subject: [OpenSIPS-Users] dbalias and location lookup branching
In-Reply-To: <92e86a5c-1d8e-4c06-af72-fd7165be971b@opensips.org>
References: <CO6P222MB0570536DEEBE3DC963F05FABE0C3A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>
 <CAEZrhoNaRfRi7L1ZGnfem+C91LTx-qPTM4pXH9Js+MLXAFcsqQ@mail.gmail.com>
 <92e86a5c-1d8e-4c06-af72-fd7165be971b@opensips.org>
Message-ID: <CO6P222MB05705B9E775B2A7A92EE07B5E0C2A@CO6P222MB0570.NAMP222.PROD.OUTLOOK.COM>

Thanks for the response Ravzan,

I attempted what I think is the mentioned solution, and apologies if this is not correct as I do not have a firm grasp of how osips handles branching, but by removing the branch route and doing location lookup in the main route right after the db_alias lookup if there are multiple sip users that dbalias resolves to then it's not looking up location for those branches, just the main branch.

### CODE ###

        alias_db_lookup("dbaliases");
        if ( !lookup("location") ) {
            sl_send_reply(404, "User not found");
            return(0);
        }

### CODE ###

So branched INVITEs are just getting routed without being looked up, which routes them back to osips. Is there a way to call the lookup in the main route and have it perform location lookups for all the branches created by dbalias lookup?

And in the previous sample that I linked with lookups called from the branch_route, calls will work as long as there's a UAS to call.  So as long as there's 1 branch that hasn't been dropped it will send that out, the only issue is if there are no possible endpoints. So if there's no sip users for that extension or no registrations for all sip users, I was trying to check the return code from t_relay and expected if there were no branches then relay would return a -3 response code but it seemingly still returns a 1 and a 500 SIP response is sent back to the UAC.  If I can get the t_relay to just return a -3 then all my problems are solved I think.  Or if there's a way to check before the relay if there are still branches to send and if not then respond 404.
________________________________
No, this is not the solution :).

The problem is you are calling t_relay, but after you evaluate each
branch, it turns out t_relay does not actually relay anything, hence it
returns an error.

The proper way to do this is to figure out whether you do need to send
any branches before calling t_relay() - this means that after the
alias_db_lookup, you can simply call the lookup() function - if that
fails, you should reply with a 404.
But what I am missing is the 500 message - what happened with the call
to the actual extension? Because from your script, it appears you still
want to keep it as a branch, don't you? Isn't that branch properly sent?

Also, the 500 sent to the client is very likely sent by the
sl_reply_error() function - you can replace it with a 404. However, I'd
still refactor everything to handle the branches in the main processing
context, not in the branch route.

My 2cents, best regards,

Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com / https://www.siphub.com

On 9/27/23 13:54, M S wrote:
> Maybe a send_reply(404) after if(!lookup(location)), instead of drop?
>
> On Tue, Sep 26, 2023 at 8:30 PM John Sliney <John.Sliney at lcs.com
> <mailto:John.Sliney at lcs.com>> wrote:
>
>     Hi,
>
>     I’m currently attempting to take an INVITE from an Asterisk server
>     that is requesting an extension number, perform dbalias lookups to
>     have extensions turned into sip users (x1000 -> test_hardphone) and
>     then do location lookups on those sip users.  There can be multiple
>     sip users for each extension and multiple locations for each sip user.
>
>     Using the code below partially works but when there are no location
>     entries for the requested sip user, OpenSIPS returns a 500 and
>     prints out “ERROR:  t_forward_nonack failed” how can I
>     have OpenSIPS instead respond with a 404?
>
>     ### CODE ###
>
>     modparam("alias_db",        "append_branches",          1)
>
>     route {
>          if ( is_from_gw() ) {
>              alias_db_lookup("dbaliases");
>              t_on_branch("sip_user_branch");
>          }
>          route(relay);
>          return(0);
>     }
>
>     branch_route[sip_user_branch] {
>          route(lookup_sip_user);
>     }
>
>     route[lookup_sip_user] {
>          if ( ! lookup("location")  ) {
>              drop();
>          }
>     }
>
>     route[relay] {
>          if ( ! t_relay() ) {
>              sl_reply_error();
>              rtpproxy_unforce();
>              return(0);
>          }
>     }
>
>     ### CODE ###
>
>     Any help would be appreciated, Thanks
>     _______________________________________________
>     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
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From 9to1url at gmail.com  Fri Sep 29 16:15:22 2023
From: 9to1url at gmail.com (Nine to one)
Date: Fri, 29 Sep 2023 12:15:22 -0400
Subject: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version
	question
Message-ID: <CAOPUfWXK24wv-XmRT-yWUucb0yH6qtqMnwTwJEHXJsozXoW0rA@mail.gmail.com>

Hello OpenSIPS Control Panel developers,

>From website OCP only mentioned support up to OpenSIPS 3.3, I am using 3.4,
so want to know if current OCP already support OpenSIPS 3.4 or not.

Thanks,
Nineto
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From efes99999 at gmail.com  Sat Sep 30 14:16:23 2023
From: efes99999 at gmail.com (L S)
Date: Sat, 30 Sep 2023 10:16:23 -0400
Subject: [OpenSIPS-Users] Can't set TLS ciphers_list to NULL
Message-ID: <CAPVhdZ_E+0ksMbF4pYOW0ZRrofbs6XrUg96NhzCw=FQq+zTexw@mail.gmail.com>

Wolfssl gives an error and Opensips doesn't start when trying to set the
ciphers_list to NULL for a client domain in 3.2.13.

modparam("tls_mgm", "ciphers_list", "[testclient]NULL")

ERROR:tls_wolfssl:_wolfssl_init_tls_dom: failure to set SSL context cipher
list 'NULL'

Any suggestions?

Thanks,
Matt
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