From marcin at voipplus.net  Tue Mar  1 00:18:28 2022
From: marcin at voipplus.net (Marcin Groszek)
Date: Mon, 28 Feb 2022 18:18:28 -0600
Subject: [OpenSIPS-Users] acc not writing INVITE to db
In-Reply-To: <CAC9cSOCsc4=7xoA6Zx=qkwYYq2xWUmWn_yd5etFGNE_WdAR79A@mail.gmail.com>
References: <5c9faee5-23ea-9780-3168-d5b3dda371de@voipplus.net>
 <8a718237-104e-50b1-06c4-0fd48cf70ecd@voipplus.net>
 <f2b60485-f397-165d-b28a-85d014900f19@opensips.org>
 <ad744458-e25c-587a-236f-edd5e818681f@voipplus.net>
 <ff2a0eda-73f8-16f2-9c68-17fdebf72773@voipplus.net>
 <2511a995-cabb-a832-f48b-853889d412ad@voipplus.net>
 <bea3a9e6-65a1-47c0-4dbf-31baf1b92793@opensips.org>
 <b4358c61-76f6-6268-3d28-9722d7899d6e@voipplus.net>
 <45160c46-a2fa-95c4-86a6-8e4b66783120@voipplus.net>
 <CAC9cSOCsc4=7xoA6Zx=qkwYYq2xWUmWn_yd5etFGNE_WdAR79A@mail.gmail.com>
Message-ID: <a52323d6-847a-4c4d-9cb1-3b6b797f9696@voipplus.net>

In my case upgrade was not relevant. The issue has been a database 
structure. We attempted to use master/master replication between 3 nodes 
that included dialog table and written dialog entry has been corrupting 
as soon as got written by a mariadb remote cluster node, It worked just 
fine with a local cluster node.

I have spent weeks to figure it out. During testing and troubleshooting 
the results ware not consistent. It would work once and not work next 
test without any changes. It has been very time consuming, and all I got 
from debug was that the dialog was not found.

Check your DBG, as Răzvan wrote : CDRs are based on dialog support, 
ngrep or log database entries if you see an attempts to be written. If 
you use extra_fields database may reject insert if the value for 
extra_field is NULL or wrong format depending on type of extra_field.

By no means I am an expert, but if your extra_field is NOT NULL and 
value is not supplied the whole insert is rejected.

Hope this helps a bit.

On 2/28/2022 12:48 AM, Saint Michael wrote:
> I have a similar issue, which probably  started with an update
> 30% of my calls' CDR do not get written to disk after
>
> OPENSIPSCTL=/usr/local/bin/opensips-cli
> /usr/bin/timeout -k 5 5 ${OPENSIPSCTL} -x mi flat_rotate 2>&1 >> 
> /usr/src/cdr.log
>
> any idea what can be happening?
>
>
>
> On Sun, Feb 27, 2022 at 4:44 PM Marcin Groszek <marcin at voipplus.net 
> <mailto:marcin at voipplus.net>> wrote:
>
>     You may disregard. I have located the issue.
>
>     -- 
>     Best Regards:
>     Marcin Groszek
>     Business Voip Resource.
>     http://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
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Best Regards:
Marcin Groszek
Business Voip Resource.
http://www.voipplus.net

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From vladp at opensips.org  Tue Mar  1 09:53:12 2022
From: vladp at opensips.org (Vlad Patrascu)
Date: Tue, 1 Mar 2022 11:53:12 +0200
Subject: [OpenSIPS-Users] is_from_gw() DNS Names
In-Reply-To: <CAOvQDeuSNjhJbUGESwgZ9M03sCSpiQPeUsR4TrL6OVhmGN3gEg@mail.gmail.com>
References: <CAOvQDetAw39bu7r=-v7PVee9pTZaa8Q6qvU0r=-Q=cx8BjGJNw@mail.gmail.com>
 <79c224ab-d62b-38ba-302c-387b1f73ec3d@opensips.org>
 <CAOvQDet1TC=0c1sq9aan804nbcF50+UJzuVy8qh_+c8pXsGrzQ@mail.gmail.com>
 <73a1d1da-64f2-d073-78d6-522c53927b57@opensips.org>
 <CAOvQDet6QMn_fopF_A91inUsNn0C2iOvUPW6=qHTS37D2scyzw@mail.gmail.com>
 <CAND0Lkv9Wo2kmwUinM5_hNbBNhLV9vqXnRHpkSYJURXeojnCUg@mail.gmail.com>
 <CAOvQDeuSNjhJbUGESwgZ9M03sCSpiQPeUsR4TrL6OVhmGN3gEg@mail.gmail.com>
Message-ID: <6ad4a846-0c09-664e-da26-3d72a5bbe50c@opensips.org>

Hi Mark,

We are aware of this limitation with wolfssl, and do plan to address it 
somehow but we have not found a straight-forward solution yet. Keep an 
eye on the feature request Ovidiu mentioned.

Regards,

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

On 28.02.2022 10:50, Mark Farmer wrote:
> Thanks Ovidiu, that is great information.
>
> I am using wolfssl as that seems to be the way to go these days.
> I wonder given the rising popularity of Direct Routing if it would be 
> possible/sensible to have wolfsssl populate the $tls_peer_subject_cn 
> variable in the future?
>
> Mark.
>
>
>
>
>
> On Fri, 25 Feb 2022 at 17:32, Ovidiu Sas <osas at voipembedded.com> wrote:
>
>     With MS, you can authenticate based on $tls_peer_subject_cn. This
>     works ok with openssl but not with wolfssl. When wolfssl is using
>     session tickets to establish new connections, the $tls_peer_subject_cn
>     is not populated.
>     Another alternative is to perform a lookup for each request received
>     over a tls connection using the ip.resolve transformation and enable
>     dbs_cache to help a little bit. It's messy but it works.
>
>     -ovidiu
>
>     On Fri, Feb 25, 2022 at 6:51 AM Mark Farmer <farmorg at gmail.com> wrote:
>     >
>     > Thanks Bogdan
>     >
>     > It's no secret really, I was just speaking generically.
>     > They are the MS Direct Routing domains, EG
>     sip.pstnhub.microsoft.com <http://sip.pstnhub.microsoft.com>
>     >
>     > Mark.
>     >
>     >
>     >
>     > On Tue, 22 Feb 2022 at 12:50, Bogdan-Andrei Iancu
>     <bogdan at opensips.org> wrote:
>     >>
>     >> Hi Mark,
>     >>
>     >> You say the DNS is publishing only one IP for the domain, but
>     one may change ? If you want, you can PM me the actual domain to
>     see how the DNS records looks like.
>     >>
>     >> Regards,
>     >>
>     >> Bogdan-Andrei Iancu
>     >>
>     >> OpenSIPS Founder and Developer
>     >> https://www.opensips-solutions.com
>     >> OpenSIPS eBootcamp
>     >> https://www.opensips.org/Training/Bootcamp
>     >>
>     >> On 2/22/22 12:31 PM, Mark Farmer wrote:
>     >>
>     >> Hi Bogdan
>     >>
>     >> The GW's have 2 CNAME records which I have no control over. DR
>     has entries like subdomain.example.com:5061
>     <http://subdomain.example.com:5061>
>     >> I suspect the issue arises when the CNAMES swap around
>     resulting in a mismatch.
>     >>
>     >> Currently I am using this to identify the source of the message
>     which is probably not the best in terms of security.
>     >>
>     >> $avp(fd) = "subdomain.example.com <http://subdomain.example.com>";
>     >> if($(ct.fields(uri){s.index, $avp(fd)}) != NULL)
>     >>
>     >> Perhaps there is a better way?
>     >>
>     >> Best regards
>     >> Mark.
>     >>
>     >>
>     >>
>     >> On Tue, 22 Feb 2022 at 08:56, Bogdan-Andrei Iancu
>     <bogdan at opensips.org> wrote:
>     >>>
>     >>> Hi Mark,
>     >>>
>     >>> If a gw is defined via FQDN, that will by DNS resolved (NAPTR,
>     SRV, A records) when DB data is (re)loaded by DR module, and used
>     later for such checks. All found IPs (from DNS) will be stored on
>     the GW.
>     >>>
>     >>> How do you specify the GW address in DB and what kind of DNS
>     records do you have for it ?
>     >>>
>     >>> Best regards,
>     >>>
>     >>> Bogdan-Andrei Iancu
>     >>>
>     >>> OpenSIPS Founder and Developer
>     >>> https://www.opensips-solutions.com
>     >>> OpenSIPS eBootcamp
>     >>> https://www.opensips.org/Training/Bootcamp
>     >>>
>     >>> On 2/18/22 6:04 PM, Mark Farmer wrote:
>     >>>
>     >>> Hi everyone
>     >>>
>     >>> I am using is_from_gw() to match against a group of gateways
>     specified by DNS names which resolve to multiple IP addresses but
>     it seems to be failing to match.
>     >>>
>     >>> Is this supported functionality or do I need to do something
>     else in this case?
>     >>>
>     >>> Thanks and regards
>     >>> Mark.
>     >>>
>     >>>
>     >>> _______________________________________________
>     >>> Users mailing list
>     >>> Users at lists.opensips.org
>     >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>     >>>
>     >>>
>     >>
>     >>
>     >> --
>     >> Mark Farmer
>     >> farmorg at gmail.com
>     >>
>     >>
>     >
>     >
>     > --
>     > Mark Farmer
>     > farmorg at gmail.com
>     > _______________________________________________
>     > Users mailing list
>     > Users at lists.opensips.org
>     > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
>     -- 
>     VoIP Embedded, Inc.
>     http://www.voipembedded.com
>
>     _______________________________________________
>     Users mailing list
>     Users at lists.opensips.org
>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> -- 
> Mark Farmer
> farmorg at gmail.com
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From razvan at opensips.org  Tue Mar  1 10:23:46 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Tue, 1 Mar 2022 12:23:46 +0200
Subject: [OpenSIPS-Users] [OpenSIPS-Devel] B2B Logic migration from XML
	to script
In-Reply-To: <ccef1873-2ef2-04df-86e2-5d132b8193dd@opensips.org>
References: <ccef1873-2ef2-04df-86e2-5d132b8193dd@opensips.org>
Message-ID: <96c2eb5e-6f89-e05d-cc6e-70492149037a@opensips.org>

Hi, Everyone!

Just a kind reminder about sharing your B2B Logic migration experience[3].
Note that after the deadline, 27th of March 2022, the b2b_logic_xml 
module will be removed.

> [3] 
> https://docs.google.com/forms/d/e/1FAIpQLScoYpSybDE5ul5zkBhsqjuLStBjXqwI7ED2BCpY3IOl0jb5Og/viewform 


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



From farmorg at gmail.com  Tue Mar  1 14:05:49 2022
From: farmorg at gmail.com (Mark Farmer)
Date: Tue, 1 Mar 2022 14:05:49 +0000
Subject: [OpenSIPS-Users] is_from_gw() DNS Names
In-Reply-To: <6ad4a846-0c09-664e-da26-3d72a5bbe50c@opensips.org>
References: <CAOvQDetAw39bu7r=-v7PVee9pTZaa8Q6qvU0r=-Q=cx8BjGJNw@mail.gmail.com>
 <79c224ab-d62b-38ba-302c-387b1f73ec3d@opensips.org>
 <CAOvQDet1TC=0c1sq9aan804nbcF50+UJzuVy8qh_+c8pXsGrzQ@mail.gmail.com>
 <73a1d1da-64f2-d073-78d6-522c53927b57@opensips.org>
 <CAOvQDet6QMn_fopF_A91inUsNn0C2iOvUPW6=qHTS37D2scyzw@mail.gmail.com>
 <CAND0Lkv9Wo2kmwUinM5_hNbBNhLV9vqXnRHpkSYJURXeojnCUg@mail.gmail.com>
 <CAOvQDeuSNjhJbUGESwgZ9M03sCSpiQPeUsR4TrL6OVhmGN3gEg@mail.gmail.com>
 <6ad4a846-0c09-664e-da26-3d72a5bbe50c@opensips.org>
Message-ID: <CAOvQDesEAZMgD8sfeQxO_=uyxdyiqo91D6=iNH8gLoF-t-k8Dg@mail.gmail.com>

Thanks both, will do.

Mark.


On Tue, 1 Mar 2022 at 09:56, Vlad Patrascu <vladp at opensips.org> wrote:

> Hi Mark,
>
> We are aware of this limitation with wolfssl, and do plan to address it
> somehow but we have not found a straight-forward solution yet. Keep an eye
> on the feature request Ovidiu mentioned.
>
> Regards,
>
> --
> Vlad Patrascu
> OpenSIPS Core Developerhttp://www.opensips-solutions.com
>
> On 28.02.2022 10:50, Mark Farmer wrote:
>
> Thanks Ovidiu, that is great information.
>
> I am using wolfssl as that seems to be the way to go these days.
> I wonder given the rising popularity of Direct Routing if it would be
> possible/sensible to have wolfsssl populate the $tls_peer_subject_cn
> variable in the future?
>
> Mark.
>
>
>
>
>
> On Fri, 25 Feb 2022 at 17:32, Ovidiu Sas <osas at voipembedded.com> wrote:
>
>> With MS, you can authenticate based on $tls_peer_subject_cn. This
>> works ok with openssl but not with wolfssl. When wolfssl is using
>> session tickets to establish new connections, the $tls_peer_subject_cn
>> is not populated.
>> Another alternative is to perform a lookup for each request received
>> over a tls connection using the ip.resolve transformation and enable
>> dbs_cache to help a little bit. It's messy but it works.
>>
>> -ovidiu
>>
>> On Fri, Feb 25, 2022 at 6:51 AM Mark Farmer <farmorg at gmail.com> wrote:
>> >
>> > Thanks Bogdan
>> >
>> > It's no secret really, I was just speaking generically.
>> > They are the MS Direct Routing domains, EG sip.pstnhub.microsoft.com
>> >
>> > Mark.
>> >
>> >
>> >
>> > On Tue, 22 Feb 2022 at 12:50, Bogdan-Andrei Iancu <bogdan at opensips.org>
>> wrote:
>> >>
>> >> Hi Mark,
>> >>
>> >> You say the DNS is publishing only one IP for the domain, but one may
>> change ? If you want, you can PM me the actual domain to see how the DNS
>> records looks like.
>> >>
>> >> Regards,
>> >>
>> >> Bogdan-Andrei Iancu
>> >>
>> >> OpenSIPS Founder and Developer
>> >>   https://www.opensips-solutions.com
>> >> OpenSIPS eBootcamp
>> >>   https://www.opensips.org/Training/Bootcamp
>> >>
>> >> On 2/22/22 12:31 PM, Mark Farmer wrote:
>> >>
>> >> Hi Bogdan
>> >>
>> >> The GW's have 2 CNAME records which I have no control over. DR has
>> entries like subdomain.example.com:5061
>> >> I suspect the issue arises when the CNAMES swap around resulting in a
>> mismatch.
>> >>
>> >> Currently I am using this to identify the source of the message which
>> is probably not the best in terms of security.
>> >>
>> >> $avp(fd) = "subdomain.example.com";
>> >> if($(ct.fields(uri){s.index, $avp(fd)}) != NULL)
>> >>
>> >> Perhaps there is a better way?
>> >>
>> >> Best regards
>> >> Mark.
>> >>
>> >>
>> >>
>> >> On Tue, 22 Feb 2022 at 08:56, Bogdan-Andrei Iancu <bogdan at opensips.org>
>> wrote:
>> >>>
>> >>> Hi Mark,
>> >>>
>> >>> If a gw is defined via FQDN, that will by DNS resolved (NAPTR, SRV, A
>> records) when DB data is (re)loaded by DR module, and used later for such
>> checks. All found IPs (from DNS) will be stored on the GW.
>> >>>
>> >>> How do you specify the GW address in DB and what kind of DNS records
>> do you have for it ?
>> >>>
>> >>> Best regards,
>> >>>
>> >>> Bogdan-Andrei Iancu
>> >>>
>> >>> OpenSIPS Founder and Developer
>> >>>   https://www.opensips-solutions.com
>> >>> OpenSIPS eBootcamp
>> >>>   https://www.opensips.org/Training/Bootcamp
>> >>>
>> >>> On 2/18/22 6:04 PM, Mark Farmer wrote:
>> >>>
>> >>> Hi everyone
>> >>>
>> >>> I am using is_from_gw() to match against a group of gateways
>> specified by DNS names which resolve to multiple IP addresses but it seems
>> to be failing to match.
>> >>>
>> >>> Is this supported functionality or do I need to do something else in
>> this case?
>> >>>
>> >>> Thanks and regards
>> >>> Mark.
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> Users mailing list
>> >>> Users at lists.opensips.org
>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >>>
>> >>>
>> >>
>> >>
>> >> --
>> >> Mark Farmer
>> >> farmorg at gmail.com
>> >>
>> >>
>> >
>> >
>> > --
>> > Mark Farmer
>> > farmorg at gmail.com
>> > _______________________________________________
>> > Users mailing list
>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>> --
>> VoIP Embedded, Inc.
>> http://www.voipembedded.com
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Mark Farmer
> farmorg at gmail.com
>
> _______________________________________________
> 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
>


-- 
Mark Farmer
farmorg at gmail.com
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From bogdan at opensips.org  Tue Mar  1 14:28:43 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 1 Mar 2022 16:28:43 +0200
Subject: [OpenSIPS-Users] OpenSIPS 3.2 + MySQL - do_accounting()
In-Reply-To: <98892702-b286-0ed3-9837-9391d313c36a@teconisy.com>
References: <98892702-b286-0ed3-9837-9391d313c36a@teconisy.com>
Message-ID: <a7c3d09a-972d-0aca-8028-31606539ea24@opensips.org>

Hi Eugen,

What is the capture of the call that produced that CDR ? Can you 
reproduce it ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp
   https://www.opensips.org/Training/Bootcamp

On 2/23/22 11:21 PM, Eugen Prieb via Users wrote:
> Hello,
>
> I will collect all CDRs, also failled, in DB.
>
> I see in log follow message:
> Feb 23 21:58:33 opensips-32 /sbin/opensips[2623292]: 
> CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error 
> (1048): Column 'to_tag' cannot be null
>
> As MySQL is MariaDB 10.5.12 installed. Have you any idee?
>
>



From bogdan at opensips.org  Tue Mar  1 14:30:18 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Tue, 1 Mar 2022 16:30:18 +0200
Subject: [OpenSIPS-Users] Attended call transfer in opensips with use of
 RTPengine
In-Reply-To: <55ddf70b-10a8-6c1f-6d04-f81f3bce311a@softnet.si>
References: <b0215761-dc49-1b60-558d-418bd216b63e@softnet.si>
 <55ddf70b-10a8-6c1f-6d04-f81f3bce311a@softnet.si>
Message-ID: <21daf27a-bfec-04d5-184c-55f1deeef1b2@opensips.org>

Hi Simon,

Do you use B2B on the OpenSIPS side ? Which entity is actually 
performing the transfer ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp
   https://www.opensips.org/Training/Bootcamp

On 2/24/22 1:54 PM, Simon Gajski via Users wrote:
>
> Hi
>
>
> I am using opensips 3.2 with rtpengine on same server and trying to 
> achieve attended call transfer.
>
> In theory, I'm trying to do:
> 1. A calls B...and B answers
> 2. B puts A on hold (MOH is played from RTPengine)
> 3. B calls C...and C answers
>
> Now the funny part:
> B tries to transfer A to C and sends REFER to opensips
> In opensips I responds with 202 Accepted and B gets disconnected.
>
> However A and C don't get connected together
> A still receives MOH and C has no voice
>
> We have another installation of opensips where REFER handles 
> Freeswitch, and there such type of transfer is working fine.
>
> Can someone help me how to handle such call behaviour in opensips with 
> RTPengine?
>
>
> relevant part of code:
>
> route[handle_sequential]{
> ...
> if(is_method("REFER")) {
>         xlog("[IN_DIALOG] [$rm] Transfer from $fu to $tu");
>         send_reply(202, "Accepted");
>
>         #what next?
>
>         exit;
>     }
> ...
> }
>
>
> Thank you!
>
> Simon
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From venefax at gmail.com  Tue Mar  1 14:54:11 2022
From: venefax at gmail.com (Saint Michael)
Date: Tue, 1 Mar 2022 09:54:11 -0500
Subject: [OpenSIPS-Users] OpenSIPS 3.2 + MySQL - do_accounting()
In-Reply-To: <a7c3d09a-972d-0aca-8028-31606539ea24@opensips.org>
References: <98892702-b286-0ed3-9837-9391d313c36a@teconisy.com>
 <a7c3d09a-972d-0aca-8028-31606539ea24@opensips.org>
Message-ID: <CAC9cSOBACD8Ldkb5NJmAatjgQNAGPWSs6BqBGz2WE5xZDWHMKA@mail.gmail.com>

In my case, Opensips is losing 30% of CDR, but there is no error.
At routing time I create a CDR record based on SIP callID, and Opensips is
supposed to "close" it when the call drops by executing a stored procedure
from the place when the call is finished. It does not in 30% of cases. I
started doing this because my carrier was showing a lot more than I did in
minutes. The CDR written to disk the traditional way also shows no record
of so many calls. In 541000 calls I have no accounting for 80.000.
nightmare.
Any idea what can it be?
Vlad is my advisor, please copy him on the information.


On Tue, Mar 1, 2022 at 9:31 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Eugen,
>
> What is the capture of the call that produced that CDR ? Can you
> reproduce it ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp
>    https://www.opensips.org/Training/Bootcamp
>
> On 2/23/22 11:21 PM, Eugen Prieb via Users wrote:
> > Hello,
> >
> > I will collect all CDRs, also failled, in DB.
> >
> > I see in log follow message:
> > Feb 23 21:58:33 opensips-32 /sbin/opensips[2623292]:
> > CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error
> > (1048): Column 'to_tag' cannot be null
> >
> > As MySQL is MariaDB 10.5.12 installed. Have you any idee?
> >
> >
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From hunterj91 at hotmail.com  Tue Mar  1 17:03:33 2022
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Tue, 1 Mar 2022 17:03:33 +0000
Subject: [OpenSIPS-Users] OpenSIPS user location clustering in Kubernetes
	behind AWS ELB
Message-ID: <AS8P195MB1287FD33FE5C11EDACB10D55BD029@AS8P195MB1287.EURP195.PROD.OUTLOOK.COM>

Hi All,

I am using openSIPS to load balance requests from Websocket clients that are using sip.js and are connecting to AWS via WSS and the ELB, the requests are then sent to freeSWITCH for some media functions and then passed back.

With a single openSIPS pod this works great, but I am now looking to implement more than one openSIPS instance, so user location then becomes an issue as I am routing using namespace.

I have used the clusterer module when not in a dynamic k8s environment, however I wondered if anyone else had tried/tested this?

For me the neighbour node Ids may be an issue due to the dynamic nature of pods, and the node_ids themselves, as I guess I can prepopulate them but wondered if this had been tested out?

Just want to make the openSIPS pods HA and scalable.

Hope this makes sense? I am looking at other work arounds but thought would put it out there.

Thanks

Jon

Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

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From simon at softnet.si  Wed Mar  2 10:02:13 2022
From: simon at softnet.si (Simon Gajski)
Date: Wed, 2 Mar 2022 11:02:13 +0100
Subject: [OpenSIPS-Users] Attended call transfer in opensips with use of
 RTPengine
In-Reply-To: <21daf27a-bfec-04d5-184c-55f1deeef1b2@opensips.org>
References: <b0215761-dc49-1b60-558d-418bd216b63e@softnet.si>
 <55ddf70b-10a8-6c1f-6d04-f81f3bce311a@softnet.si>
 <21daf27a-bfec-04d5-184c-55f1deeef1b2@opensips.org>
Message-ID: <f0fb2d68-6a0e-d3a2-b940-55d84e01c92b@softnet.si>

Hi

no, we don't use B2B on OpenSIPS  side. Is this the correctway to do it?
The thing is that I don't know what would be the best way to implement 
this with use of RTPengine.
I found very little info available online.

Call forwards that I manually set in DB (cfu, cfnr, cfb.....like we were 
doing on last bootcamp) are working fine.
Only issue is with answered call and then attempting to transfer it.

BR
Simon



Bogdan-Andrei Iancu je 01.03.2022 ob 15:30 napisal:
> Hi Simon,
>
> Do you use B2B on the OpenSIPS side ? Which entity is actually 
> performing the transfer ?
>
> Regards,
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp
>    https://www.opensips.org/Training/Bootcamp
> On 2/24/22 1:54 PM, Simon Gajski via Users wrote:
>>
>> Hi
>>
>>
>> I am using opensips 3.2 with rtpengine on same server and trying to 
>> achieve attended call transfer.
>>
>> In theory, I'm trying to do:
>> 1. A calls B...and B answers
>> 2. B puts A on hold (MOH is played from RTPengine)
>> 3. B calls C...and C answers
>>
>> Now the funny part:
>> B tries to transfer A to C and sends REFER to opensips
>> In opensips I responds with 202 Accepted and B gets disconnected.
>>
>> However A and C don't get connected together
>> A still receives MOH and C has no voice
>>
>> We have another installation of opensips where REFER handles 
>> Freeswitch, and there such type of transfer is working fine.
>>
>> Can someone help me how to handle such call behaviour in opensips 
>> with RTPengine?
>>
>>
>> relevant part of code:
>>
>> route[handle_sequential]{
>> ...
>> if(is_method("REFER")) {
>>         xlog("[IN_DIALOG] [$rm] Transfer from $fu to $tu");
>>         send_reply(202, "Accepted");
>>
>>         #what next?
>>
>>         exit;
>>     }
>> ...
>> }
>>
>>
>> Thank you!
>>
>> Simon
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From callum.guy at x-on.co.uk  Thu Mar  3 17:12:06 2022
From: callum.guy at x-on.co.uk (Callum Guy)
Date: Thu, 3 Mar 2022 17:12:06 +0000
Subject: [OpenSIPS-Users] Event on DE-REGISTRATION
Message-ID: <CAFjCFzkLjP76C00Eao-F3TokcsQMP49Zm10LwK5sMdp9LcBDoA@mail.gmail.com>

Hi All,

I have configured my registrar with max_contacts 1, allowing
subsequent registrations from that contact to overwrite.

I am looking to intercept the de-registration and send a message to the
losing contact. Ideally I would use the existing SIP connection to send it
before termination, the client devices are softphones under my control, but
I suppose the TCP connection to the previous device may have been severed
by the time the event is raised (I can test this when I get there but
advice is always welcome).

Event capture is straightforward using E_UL_CONTACT_DELETE however I am
wondering if there is any way to generate SIP towards the leaving contact
at this point? t_new_request() is not available in event_route and I wanted
to reach out to the community for ideas before building an external system
to do this (i.e. rest_post to a simple HTTP client).

Is there a neat way to do this within the config script or am I already on
the best-fit path here?

Many thanks,

Callum

-- 






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From marcin at voipplus.net  Sun Mar  6 16:20:19 2022
From: marcin at voipplus.net (Marcin Groszek)
Date: Sun, 6 Mar 2022 10:20:19 -0600
Subject: [OpenSIPS-Users] suggested sql database structure
Message-ID: <48adf4d4-1c46-aa7d-ca19-fd531c38d417@voipplus.net>

     I would like to deploy opensips in multiple geographical locations 
and wondering if anyone has any suggestions on sql database structure.

dialog would be local to each instant/pop
dr_* tables are read only with possibility of dr_gateways probing
permissions read only
acc write only
avpops read/write.

Ideally I would like to see individual pops continue to operate even if 
routing to other locations is interrupted.
should I use a single centralized database , or perhaps a master/master 
cluster over public internet; testing of it was not very successful.

Any suggestions would be appreciated.


-- 
Best Regards:
Marcin Groszek
Business Voip Resource.
http://www.voipplus.net



From vinayak.makwana at ecosmob.com  Tue Mar  8 10:41:12 2022
From: vinayak.makwana at ecosmob.com (Vinayak Makwana)
Date: Tue, 8 Mar 2022 16:11:12 +0530
Subject: [OpenSIPS-Users] Header manipulation
Message-ID: <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA@mail.gmail.com>

Hello All,

I want to replace uri-display in the FROM header using the avp_subst
function. So, can anyone suggest a solution ?

Here's My scenario:
Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
After Changes ->  From:"pqrs"<sip:username at x.x.x.x
;transport=UDP>;tag=6a8eda3f

Many Thanks
Vinayak Makwana

-- 
*Disclaimer*
In addition to generic Disclaimer which you have agreed on our 
website, any views or opinions presented in this email are solely those of 
the originator and do not necessarily represent those of the Company or its 
sister concerns. Any liability (in negligence, contract or otherwise) 
arising from any third party taking any action, or refraining from taking 
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is hereby excluded.



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This communication (including any 
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From alain.bieuzent at free.fr  Tue Mar  8 10:51:45 2022
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 08 Mar 2022 11:51:45 +0100
Subject: [OpenSIPS-Users] Header manipulation
In-Reply-To: <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA@mail.gmail.com>
References: <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA@mail.gmail.com>
Message-ID: <FA2BF751-1E38-4780-AC7D-682CFE3384B4@free.fr>

Hi Vinayak,

 

Try with uac_replace_from([display],uri)

 

Ragards

 

De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 8 mars 2022 à 11:43
À : <users at lists.opensips.org>
Objet : [OpenSIPS-Users] Header manipulation

 

Hello All,

I want to replace uri-display in the FROM header using the avp_subst function. So, can anyone suggest a solution ?

Here's My scenario:
Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f 
After Changes ->  From:"pqrs"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f

Many Thanks
Vinayak Makwana

 

Disclaimer

In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.

 

Caution for viruses, malware etc.

This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. 

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

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From vinayak.makwana at ecosmob.com  Tue Mar  8 12:58:46 2022
From: vinayak.makwana at ecosmob.com (Vinayak Makwana)
Date: Tue, 8 Mar 2022 18:28:46 +0530
Subject: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
In-Reply-To: <mailman.1.1646740802.19591.users@lists.opensips.org>
References: <mailman.1.1646740802.19591.users@lists.opensips.org>
Message-ID: <CAPHmyfw4L4HiNWKgoYErYkYC60M+y7TNPa7AHAL2x7CFfVTCyg@mail.gmail.com>

Hello Alain Bieuzent

I tried with the uac_replace_from() function also but not getting the
proper result.

Here's my input & output result:
INPUT:      From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
OUTPUT:  From:abc<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f

Here's my logic:
$avp(ds)="abc";
        uac_replace_from($avp(ds),"");

So Can you please tell me what is an issue why not getting quotes

Thanks in advance
Vinayak Makwana

On Tue, Mar 8, 2022 at 5:30 PM <users-request at lists.opensips.org> wrote:

> Send Users mailing list submissions to
>         users at lists.opensips.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> or, via email, send a message with subject or body 'help' to
>         users-request at lists.opensips.org
>
> You can reach the person managing the list at
>         users-owner at lists.opensips.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Users digest..."
>
>
> Today's Topics:
>
>    1. Header manipulation (Vinayak Makwana)
>    2. Re: Header manipulation (Alain Bieuzent)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Tue, 8 Mar 2022 16:11:12 +0530
> From: Vinayak Makwana <vinayak.makwana at ecosmob.com>
> To: users at lists.opensips.org
> Subject: [OpenSIPS-Users] Header manipulation
> Message-ID:
>         <
> CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hello All,
>
> I want to replace uri-display in the FROM header using the avp_subst
> function. So, can anyone suggest a solution ?
>
> Here's My scenario:
> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
> After Changes ->  From:"pqrs"<sip:username at x.x.x.x
> ;transport=UDP>;tag=6a8eda3f
>
> Many Thanks
> Vinayak Makwana
>
> --
> *Disclaimer*
> In addition to generic Disclaimer which you have agreed on our
> website, any views or opinions presented in this email are solely those of
> the originator and do not necessarily represent those of the Company or
> its
> sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and
> contains
> information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading,
> dissemination, distribution, or copying of this communication is
> prohibited. Please inform originator if you have received it in error.
>
>
> *Caution for viruses, malware etc.*
> This communication, including any
> attachments, may not be free of viruses, trojans, similar or new
> contaminants/malware, interceptions or interference, and may not be
> compatible with your systems. You shall carry out virus/malware scanning
> on
> your own before opening any attachment to this e-mail. The sender of this
> e-mail and Company including its sister concerns shall not be liable for
> any damage that may incur to you as a result of viruses, incompleteness of
> this message, a delay in receipt of this message or any other computer
> problems.
> -------------- next part --------------
> An HTML attachment was scrubbed...
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> http://lists.opensips.org/pipermail/users/attachments/20220308/d383bf19/attachment-0001.html
> >
>
> ------------------------------
>
> Message: 2
> Date: Tue, 08 Mar 2022 11:51:45 +0100
> From: Alain Bieuzent <alain.bieuzent at free.fr>
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Subject: Re: [OpenSIPS-Users] Header manipulation
> Message-ID: <FA2BF751-1E38-4780-AC7D-682CFE3384B4 at free.fr>
> Content-Type: text/plain; charset="utf-8"
>
> Hi Vinayak,
>
>
>
> Try with uac_replace_from([display],uri)
>
>
>
> Ragards
>
>
>
> De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <
> vinayak.makwana at ecosmob.com>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
> Date : mardi 8 mars 2022 à 11:43
> À : <users at lists.opensips.org>
> Objet : [OpenSIPS-Users] Header manipulation
>
>
>
> Hello All,
>
> I want to replace uri-display in the FROM header using the avp_subst
> function. So, can anyone suggest a solution ?
>
> Here's My scenario:
> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
> After Changes ->  From:"pqrs"<sip:username at x.x.x.x
> ;transport=UDP>;tag=6a8eda3f
>
> Many Thanks
> Vinayak Makwana
>
>
>
> Disclaimer
>
> In addition to generic Disclaimer which you have agreed on our website,
> any views or opinions presented in this email are solely those of the
> originator and do not necessarily represent those of the Company or its
> sister concerns. Any liability (in negligence, contract 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 the
> use of the addressee(s) and contains information that is PRIVILEGED AND
> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying
> of this communication is prohibited. Please inform originator if you have
> received it in error.
>
>
>
> Caution for viruses, malware etc.
>
> This communication, including any attachments, may not be free of viruses,
> trojans, similar or new contaminants/malware, interceptions or
> interference, and may not be compatible with your systems. You shall carry
> out virus/malware scanning on your own before opening any attachment to
> this e-mail. The sender of this e-mail and Company including its sister
> concerns shall not be liable for any damage that may incur to you as a
> result of viruses, incompleteness of this message, a delay in receipt of
> this message or any other computer problems.
>
> _______________________________________________ Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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> ------------------------------
>
> Subject: Digest Footer
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
> ------------------------------
>
> End of Users Digest, Vol 164, Issue 6
> *************************************
>

-- 
*Disclaimer*
In addition to generic Disclaimer which you have agreed on our 
website, any views or opinions presented in this email are solely those of 
the originator and do not necessarily represent those of the Company or its 
sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains 
information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, 
dissemination, distribution, or copying of this communication is 
prohibited. Please inform originator if you have received it in error.


*Caution for viruses, malware etc.*
This communication, including any 
attachments, may not be free of viruses, trojans, similar or new 
contaminants/malware, interceptions or interference, and may not be 
compatible with your systems. You shall carry out virus/malware scanning on 
your own before opening any attachment to this e-mail. The sender of this 
e-mail and Company including its sister concerns shall not be liable for 
any damage that may incur to you as a result of viruses, incompleteness of 
this message, a delay in receipt of this message or any other computer 
problems. 
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From alain.bieuzent at free.fr  Tue Mar  8 14:12:34 2022
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 08 Mar 2022 15:12:34 +0100
Subject: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
In-Reply-To: <CAPHmyfw4L4HiNWKgoYErYkYC60M+y7TNPa7AHAL2x7CFfVTCyg@mail.gmail.com>
References: <mailman.1.1646740802.19591.users@lists.opensips.org>
 <CAPHmyfw4L4HiNWKgoYErYkYC60M+y7TNPa7AHAL2x7CFfVTCyg@mail.gmail.com>
Message-ID: <3B68F47F-33C3-4547-85AA-5E18147EA91F@free.fr>

Hmm , not sure $avp is supported can you try with $var

 

$var(ds)="abc";

        uac_replace_from($var(ds),""); 


 

De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 8 mars 2022 à 14:01
À : <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6

 

Hello Alain Bieuzent

I tried with the uac_replace_from() function also but not getting the proper result.

Here's my input & output result:
INPUT:      From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
OUTPUT:  From:abc<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f

Here's my logic: 
$avp(ds)="abc";
        uac_replace_from($avp(ds),""); 

So Can you please tell me what is an issue why not getting quotes

Thanks in advance 
Vinayak Makwana

 

On Tue, Mar 8, 2022 at 5:30 PM <users-request at lists.opensips.org> wrote:

Send Users mailing list submissions to
        users at lists.opensips.org

To subscribe or unsubscribe via the World Wide Web, visit
        http://lists.opensips.org/cgi-bin/mailman/listinfo/users
or, via email, send a message with subject or body 'help' to
        users-request at lists.opensips.org

You can reach the person managing the list at
        users-owner at lists.opensips.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of Users digest..."


Today's Topics:

   1. Header manipulation (Vinayak Makwana)
   2. Re: Header manipulation (Alain Bieuzent)


----------------------------------------------------------------------

Message: 1
Date: Tue, 8 Mar 2022 16:11:12 +0530
From: Vinayak Makwana <vinayak.makwana at ecosmob.com>
To: users at lists.opensips.org
Subject: [OpenSIPS-Users] Header manipulation
Message-ID:
        <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Hello All,

I want to replace uri-display in the FROM header using the avp_subst
function. So, can anyone suggest a solution ?

Here's My scenario:
Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
After Changes ->  From:"pqrs"<sip:username at x.x.x.x
;transport=UDP>;tag=6a8eda3f

Many Thanks
Vinayak Makwana

-- 
*Disclaimer*
In addition to generic Disclaimer which you have agreed on our 
website, any views or opinions presented in this email are solely those of 
the originator and do not necessarily represent those of the Company or its 
sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains 
information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, 
dissemination, distribution, or copying of this communication is 
prohibited. Please inform originator if you have received it in error.


*Caution for viruses, malware etc.*
This communication, including any 
attachments, may not be free of viruses, trojans, similar or new 
contaminants/malware, interceptions or interference, and may not be 
compatible with your systems. You shall carry out virus/malware scanning on 
your own before opening any attachment to this e-mail. The sender of this 
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------------------------------

Message: 2
Date: Tue, 08 Mar 2022 11:51:45 +0100
From: Alain Bieuzent <alain.bieuzent at free.fr>
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Header manipulation
Message-ID: <FA2BF751-1E38-4780-AC7D-682CFE3384B4 at free.fr>
Content-Type: text/plain; charset="utf-8"

Hi Vinayak,



Try with uac_replace_from([display],uri)



Ragards



De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 8 mars 2022 à 11:43
À : <users at lists.opensips.org>
Objet : [OpenSIPS-Users] Header manipulation



Hello All,

I want to replace uri-display in the FROM header using the avp_subst function. So, can anyone suggest a solution ?

Here's My scenario:
Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f 
After Changes ->  From:"pqrs"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f

Many Thanks
Vinayak Makwana



Disclaimer

In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.



Caution for viruses, malware etc.

This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. 

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

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------------------------------

Subject: Digest Footer

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


------------------------------

End of Users Digest, Vol 164, Issue 6
*************************************

 

Disclaimer

In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.

 

Caution for viruses, malware etc.

This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. 

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

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From alexanderhenryperkins at gmail.com  Tue Mar  8 14:28:11 2022
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Tue, 8 Mar 2022 09:28:11 -0500
Subject: [OpenSIPS-Users] Replacing From Number
Message-ID: <CALLkTp2hdWvWRT5REMvpHqS2cMesxAFKTjDRS=x3oDepVDHr8A@mail.gmail.com>

Hi All.  I am trying to replace the From Number to match that of our main
office line, but I do not seem to be having luck with it.  Here's what I am
doing:

This is from the PCAP - From: 121255551212 <sip: 121255551212 at X.X.X.X
>;tag=85698821

This is the part of my script I am using to try to replace it:

replace_body_all('$fU','19905551212');

However, the From Number is never replaced.  What should I do?

Thank you,
Alex
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From alain.bieuzent at free.fr  Tue Mar  8 14:37:41 2022
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 08 Mar 2022 15:37:41 +0100
Subject: [OpenSIPS-Users] Replacing From Number
In-Reply-To: <CALLkTp2hdWvWRT5REMvpHqS2cMesxAFKTjDRS=x3oDepVDHr8A@mail.gmail.com>
References: <CALLkTp2hdWvWRT5REMvpHqS2cMesxAFKTjDRS=x3oDepVDHr8A@mail.gmail.com>
Message-ID: <460C85F4-B0BE-44E6-BDFE-8540AAC2E58B@free.fr>

Hi Alex,

 

This the day of rewriting from ….

 

Try with uac_replace_from([display],uri)

 

Regards

 

De : Users <users-bounces at lists.opensips.org> au nom de Alexander Perkins <alexanderhenryperkins at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 8 mars 2022 à 15:30
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : [OpenSIPS-Users] Replacing From Number

 

Hi All.  I am trying to replace the From Number to match that of our main office line, but I do not seem to be having luck with it.  Here's what I am doing:

 

This is from the PCAP - From: 121255551212 <sip: 121255551212 at X.X.X.X>;tag=85698821

 

This is the part of my script I am using to try to replace it:

 

replace_body_all('$fU','19905551212');

 

However, the From Number is never replaced.  What should I do?

 

Thank you,

Alex

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

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From Johan at democon.be  Tue Mar  8 14:39:50 2022
From: Johan at democon.be (Johan De Clercq)
Date: Tue, 8 Mar 2022 15:39:50 +0100
Subject: [OpenSIPS-Users] Replacing From Number
In-Reply-To: <460C85F4-B0BE-44E6-BDFE-8540AAC2E58B@free.fr>
References: <CALLkTp2hdWvWRT5REMvpHqS2cMesxAFKTjDRS=x3oDepVDHr8A@mail.gmail.com>
 <460C85F4-B0BE-44E6-BDFE-8540AAC2E58B@free.fr>
Message-ID: <CAEVWGm8=qvDax46r6Kc8zqxxqpN31=YUT=a4K+1mzDTEjCM0dQ@mail.gmail.com>

;-)

On Tue, Mar 8, 2022, 15:39 Alain Bieuzent <alain.bieuzent at free.fr> wrote:

> Hi Alex,
>
>
>
> This the day of rewriting from ….
>
>
>
> Try with uac_replace_from([display],uri)
>
>
>
> Regards
>
>
>
> *De : *Users <users-bounces at lists.opensips.org> au nom de Alexander
> Perkins <alexanderhenryperkins at gmail.com>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *mardi 8 mars 2022 à 15:30
> *À : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Objet : *[OpenSIPS-Users] Replacing From Number
>
>
>
> Hi All.  I am trying to replace the From Number to match that of our main
> office line, but I do not seem to be having luck with it.  Here's what I am
> doing:
>
>
>
> This is from the PCAP - From: 121255551212 <sip: 121255551212 at X.X.X.X
> >;tag=85698821
>
>
>
> This is the part of my script I am using to try to replace it:
>
>
>
> replace_body_all('$fU','19905551212');
>
>
>
> However, the From Number is never replaced.  What should I do?
>
>
>
> Thank you,
>
> Alex
>
> _______________________________________________ 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 spanda at 3clogic.com  Wed Mar  9 09:53:01 2022
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 9 Mar 2022 15:23:01 +0530
Subject: [OpenSIPS-Users] Need some help in adding custome header in Cancel
	Request .
Message-ID: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>

Hi All,

Cancel is generated Hop by Hop . When the Opensips server receives a Cancel
, Then it generates Cancel for the next party .

I am adding a custom header in the Cancel request , but when the next Hop
Cancel is getting generated that custom header is not getting added . How
will I pass the custom header in the Cancel request to the destination ?

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From nick at altmann.pro  Wed Mar  9 09:57:28 2022
From: nick at altmann.pro (Nick Altmann)
Date: Wed, 9 Mar 2022 10:57:28 +0100
Subject: [OpenSIPS-Users] Need some help in adding custome header in
 Cancel Request .
In-Reply-To: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>
References: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>
Message-ID: <CADKNvCp2caOj0n4CZ+Wyt4QFbww7-0BWMJMpORZ+LYBF_K8QKw@mail.gmail.com>

Hi,

If cancel request generated by opensips, then you can control it from
local_route.

--
Nick

ср, 9 мар. 2022 г. в 10:54, Sasmita Panda <spanda at 3clogic.com>:

> Hi All,
>
> Cancel is generated Hop by Hop . When the Opensips server receives a
> Cancel , Then it generates Cancel for the next party .
>
> I am adding a custom header in the Cancel request , but when the next Hop
> Cancel is getting generated that custom header is not getting added . How
> will I pass the custom header in the Cancel request to the destination ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From spanda at 3clogic.com  Wed Mar  9 10:07:52 2022
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 9 Mar 2022 15:37:52 +0530
Subject: [OpenSIPS-Users] Need some help in adding custome header in
 Cancel Request .
In-Reply-To: <CADKNvCp2caOj0n4CZ+Wyt4QFbww7-0BWMJMpORZ+LYBF_K8QKw@mail.gmail.com>
References: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>
 <CADKNvCp2caOj0n4CZ+Wyt4QFbww7-0BWMJMpORZ+LYBF_K8QKw@mail.gmail.com>
Message-ID: <CALf-Q47=v9Eu7xsG-f7=d9rZBJ92+b=AZ_L+kKqGTzCj-dU3UQ@mail.gmail.com>

My call flow is like  below .

A -- > INVITE TO OPENSIPS -- > B
A -- > CANCEL TO OPENSIPS -- > B

While A sends Cancel to Opensips (adds a custom header ) . When Opensips
generates Cancel for B it won't add the custom header .

This can be done by local_route ?

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


On Wed, Mar 9, 2022 at 3:27 PM Nick Altmann <nick at altmann.pro> wrote:

> Hi,
>
> If cancel request generated by opensips, then you can control it from
> local_route.
>
> --
> Nick
>
> ср, 9 мар. 2022 г. в 10:54, Sasmita Panda <spanda at 3clogic.com>:
>
>> Hi All,
>>
>> Cancel is generated Hop by Hop . When the Opensips server receives a
>> Cancel , Then it generates Cancel for the next party .
>>
>> I am adding a custom header in the Cancel request , but when the next Hop
>> Cancel is getting generated that custom header is not getting added . How
>> will I pass the custom header in the Cancel request to the destination ?
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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From razvan at opensips.org  Wed Mar  9 16:14:05 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 9 Mar 2022 18:14:05 +0200
Subject: [OpenSIPS-Users] Need some help in adding custome header in
 Cancel Request .
In-Reply-To: <CALf-Q47=v9Eu7xsG-f7=d9rZBJ92+b=AZ_L+kKqGTzCj-dU3UQ@mail.gmail.com>
References: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>
 <CADKNvCp2caOj0n4CZ+Wyt4QFbww7-0BWMJMpORZ+LYBF_K8QKw@mail.gmail.com>
 <CALf-Q47=v9Eu7xsG-f7=d9rZBJ92+b=AZ_L+kKqGTzCj-dU3UQ@mail.gmail.com>
Message-ID: <16d51ba6-aab2-53e8-ff87-9e3566a632bf@opensips.org>

Hi, Sasmita!

I actually don't think local_route is run for CANCEL messages.
You may want to try to add a more complex reason using 
t_add_cancel_reason[1].

[1] https://opensips.org/docs/modules/3.2.x/tm.html#idp6205808

Best regards,

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

On 3/9/22 12:07, Sasmita Panda wrote:
> My call flow is like  below .
> 
> A -- > INVITE TO OPENSIPS -- > B
> A -- > CANCEL TO OPENSIPS -- > B
> 
> While A sends Cancel to Opensips (adds a custom header ) . When Opensips 
> generates Cancel for B it won't add the custom header .
> 
> This can be done by local_route ?
> 
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
> 
> 
> On Wed, Mar 9, 2022 at 3:27 PM Nick Altmann <nick at altmann.pro 
> <mailto:nick at altmann.pro>> wrote:
> 
>     Hi,
> 
>     If cancel request generated by opensips, then you can control it
>     from local_route.
> 
>     --
>     Nick
> 
>     ср, 9 мар. 2022 г. в 10:54, Sasmita Panda <spanda at 3clogic.com
>     <mailto:spanda at 3clogic.com>>:
> 
>         Hi All,
> 
>         Cancel is generated Hop by Hop . When the Opensips server
>         receives a Cancel , Then it generates Cancel for the next party .
> 
>         I am adding a custom header in the Cancel request , but when the
>         next Hop Cancel is getting generated that custom header is not
>         getting added . How will I pass the custom header in the Cancel
>         request to the destination ?
> 
>         */Thanks & Regards/*
>         /Sasmita Panda/
>         /Senior Network Testing and Software Engineer/
>         /3CLogic , ph:07827611765/
>         _______________________________________________
>         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 kworm at missouri-telecom.com  Wed Mar  9 18:25:36 2022
From: kworm at missouri-telecom.com (Kevin Wormington)
Date: Wed, 9 Mar 2022 12:25:36 -0600
Subject: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
In-Reply-To: <3B68F47F-33C3-4547-85AA-5E18147EA91F@free.fr>
References: <mailman.1.1646740802.19591.users@lists.opensips.org>
 <CAPHmyfw4L4HiNWKgoYErYkYC60M+y7TNPa7AHAL2x7CFfVTCyg@mail.gmail.com>
 <3B68F47F-33C3-4547-85AA-5E18147EA91F@free.fr>
Message-ID: <A2DEF8FF-9E33-4DB3-A047-57CAF180AA94@missouri-telecom.com>

AVP is for sure supported on 3.1.x and up.  We use the following:

uac_replace_from($avp(caller_cnam),"”);

> On Mar 8, 2022, at 8:12 AM, Alain Bieuzent <alain.bieuzent at free.fr> wrote:
> 
> Hmm , not sure $avp is supported can you try with $var
>  
> $var(ds)="abc";
>         uac_replace_from($var(ds),""); 
> 
>  
> De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
> Date : mardi 8 mars 2022 à 14:01
> À : <users at lists.opensips.org>
> Objet : Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
>  
> Hello Alain Bieuzent
> 
> I tried with the uac_replace_from() function also but not getting the proper result.
> 
> Here's my input & output result:
> INPUT:      From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
> OUTPUT:  From:abc<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
> 
> Here's my logic: 
> $avp(ds)="abc";
>         uac_replace_from($avp(ds),""); 
> 
> So Can you please tell me what is an issue why not getting quotes
> 
> Thanks in advance 
> Vinayak Makwana
>  
> On Tue, Mar 8, 2022 at 5:30 PM <users-request at lists.opensips.org> wrote:
>> Send Users mailing list submissions to
>>         users at lists.opensips.org
>> 
>> To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> or, via email, send a message with subject or body 'help' to
>>         users-request at lists.opensips.org
>> 
>> You can reach the person managing the list at
>>         users-owner at lists.opensips.org
>> 
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of Users digest..."
>> 
>> 
>> Today's Topics:
>> 
>>    1. Header manipulation (Vinayak Makwana)
>>    2. Re: Header manipulation (Alain Bieuzent)
>> 
>> 
>> ----------------------------------------------------------------------
>> 
>> Message: 1
>> Date: Tue, 8 Mar 2022 16:11:12 +0530
>> From: Vinayak Makwana <vinayak.makwana at ecosmob.com>
>> To: users at lists.opensips.org
>> Subject: [OpenSIPS-Users] Header manipulation
>> Message-ID:
>>         <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA at mail.gmail.com>
>> Content-Type: text/plain; charset="utf-8"
>> 
>> Hello All,
>> 
>> I want to replace uri-display in the FROM header using the avp_subst
>> function. So, can anyone suggest a solution ?
>> 
>> Here's My scenario:
>> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>> After Changes ->  From:"pqrs"<sip:username at x.x.x.x
>> ;transport=UDP>;tag=6a8eda3f
>> 
>> Many Thanks
>> Vinayak Makwana
>> 
>> -- 
>> *Disclaimer*
>> In addition to generic Disclaimer which you have agreed on our 
>> website, any views or opinions presented in this email are solely those of 
>> the originator and do not necessarily represent those of the Company or its 
>> sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains 
>> information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, 
>> dissemination, distribution, or copying of this communication is 
>> prohibited. Please inform originator if you have received it in error.
>> 
>> 
>> *Caution for viruses, malware etc.*
>> This communication, including any 
>> attachments, may not be free of viruses, trojans, similar or new 
>> contaminants/malware, interceptions or interference, and may not be 
>> compatible with your systems. You shall carry out virus/malware scanning on 
>> your own before opening any attachment to this e-mail. The sender of this 
>> e-mail and Company including its sister concerns shall not be liable for 
>> any damage that may incur to you as a result of viruses, incompleteness of 
>> this message, a delay in receipt of this message or any other computer 
>> problems. 
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: <http://lists.opensips.org/pipermail/users/attachments/20220308/d383bf19/attachment-0001.html>
>> 
>> ------------------------------
>> 
>> Message: 2
>> Date: Tue, 08 Mar 2022 11:51:45 +0100
>> From: Alain Bieuzent <alain.bieuzent at free.fr>
>> To: OpenSIPS users mailling list <users at lists.opensips.org>
>> Subject: Re: [OpenSIPS-Users] Header manipulation
>> Message-ID: <FA2BF751-1E38-4780-AC7D-682CFE3384B4 at free.fr>
>> Content-Type: text/plain; charset="utf-8"
>> 
>> Hi Vinayak,
>> 
>> 
>> 
>> Try with uac_replace_from([display],uri)
>> 
>> 
>> 
>> Ragards
>> 
>> 
>> 
>> De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
>> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
>> Date : mardi 8 mars 2022 à 11:43
>> À : <users at lists.opensips.org>
>> Objet : [OpenSIPS-Users] Header manipulation
>> 
>> 
>> 
>> Hello All,
>> 
>> I want to replace uri-display in the FROM header using the avp_subst function. So, can anyone suggest a solution ?
>> 
>> Here's My scenario:
>> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f 
>> After Changes ->  From:"pqrs"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>> 
>> Many Thanks
>> Vinayak Makwana
>> 
>> 
>> 
>> Disclaimer
>> 
>> In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.
>> 
>> 
>> 
>> Caution for viruses, malware etc.
>> 
>> This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. 
>> 
>> _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users 
>> 
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>> 
>> ------------------------------
>> 
>> Subject: Digest Footer
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> 
>> 
>> ------------------------------
>> 
>> End of Users Digest, Vol 164, Issue 6
>> *************************************
>  
> Disclaimer
> In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.
>  
> Caution for viruses, malware etc.
> This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. 
> _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> _______________________________________________
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From Ben.Newlin at genesys.com  Wed Mar  9 19:08:33 2022
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Wed, 9 Mar 2022 19:08:33 +0000
Subject: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
In-Reply-To: <A2DEF8FF-9E33-4DB3-A047-57CAF180AA94@missouri-telecom.com>
References: <mailman.1.1646740802.19591.users@lists.opensips.org>
 <CAPHmyfw4L4HiNWKgoYErYkYC60M+y7TNPa7AHAL2x7CFfVTCyg@mail.gmail.com>
 <3B68F47F-33C3-4547-85AA-5E18147EA91F@free.fr>
 <A2DEF8FF-9E33-4DB3-A047-57CAF180AA94@missouri-telecom.com>
Message-ID: <BN0PR10MB52082D777F917ED60F637469950A9@BN0PR10MB5208.namprd10.prod.outlook.com>

My guess is the lack of quoting is due to the SIP stack and not anything you are doing with your commands. The Name portion of a Name-Addr spec is only required to be quoted if certain special characters (most notably spaces) are present. Otherwise the quotes are not required. I’m guessing OpenSIPS’ SIP stack is just being efficient and only adding quotes when required. I’m not sure if that is configurable, but I’m not aware of any setting for it.

My guess is that if you were to try this:

$avp(ds)="a b c";
uac_replace_from($avp(ds),"");

You would find that the resulting output name would have quotes as required.

However, it should not matter as if quotes are not required then the receiving SIP agent should be fine with them not being present. May I ask why it is important to you that the quotes be present? Is it causing an issue with some non-compliant device?

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Kevin Wormington <kworm at missouri-telecom.com>
Date: Wednesday, March 9, 2022 at 1:26 PM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
AVP is for sure supported on 3.1.x and up.  We use the following:

uac_replace_from($avp(caller_cnam),"”);

> On Mar 8, 2022, at 8:12 AM, Alain Bieuzent <alain.bieuzent at free.fr> wrote:
>
> Hmm , not sure $avp is supported can you try with $var
>
> $var(ds)="abc";
>         uac_replace_from($var(ds),"");
>
>
> De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
> Date : mardi 8 mars 2022 à 14:01
> À : <users at lists.opensips.org>
> Objet : Re: [OpenSIPS-Users] Users Digest, Vol 164, Issue 6
>
> Hello Alain Bieuzent
>
> I tried with the uac_replace_from() function also but not getting the proper result.
>
> Here's my input & output result:
> INPUT:      From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
> OUTPUT:  From:abc<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>
> Here's my logic:
> $avp(ds)="abc";
>         uac_replace_from($avp(ds),"");
>
> So Can you please tell me what is an issue why not getting quotes
>
> Thanks in advance
> Vinayak Makwana
>
> On Tue, Mar 8, 2022 at 5:30 PM <users-request at lists.opensips.org> wrote:
>> Send Users mailing list submissions to
>>         users at lists.opensips.org
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> or, via email, send a message with subject or body 'help' to
>>         users-request at lists.opensips.org
>>
>> You can reach the person managing the list at
>>         users-owner at lists.opensips.org
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of Users digest..."
>>
>>
>> Today's Topics:
>>
>>    1. Header manipulation (Vinayak Makwana)
>>    2. Re: Header manipulation (Alain Bieuzent)
>>
>>
>> ----------------------------------------------------------------------
>>
>> Message: 1
>> Date: Tue, 8 Mar 2022 16:11:12 +0530
>> From: Vinayak Makwana <vinayak.makwana at ecosmob.com>
>> To: users at lists.opensips.org
>> Subject: [OpenSIPS-Users] Header manipulation
>> Message-ID:
>>         <CAPHmyfzdYGNoJdxX1FFjfLxOZeGQUnuAPxbAv+t8H2YX+8qXCA at mail.gmail.com>
>> Content-Type: text/plain; charset="utf-8"
>>
>> Hello All,
>>
>> I want to replace uri-display in the FROM header using the avp_subst
>> function. So, can anyone suggest a solution ?
>>
>> Here's My scenario:
>> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>> After Changes ->  From:"pqrs"<sip:username at x.x.x.x
>> ;transport=UDP>;tag=6a8eda3f
>>
>> Many Thanks
>> Vinayak Makwana
>>
>> --
>> *Disclaimer*
>> In addition to generic Disclaimer which you have agreed on our
>> website, any views or opinions presented in this email are solely those of
>> the originator and do not necessarily represent those of the Company or its
>> sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains
>> information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading,
>> dissemination, distribution, or copying of this communication is
>> prohibited. Please inform originator if you have received it in error.
>>
>>
>> *Caution for viruses, malware etc.*
>> This communication, including any
>> attachments, may not be free of viruses, trojans, similar or new
>> contaminants/malware, interceptions or interference, and may not be
>> compatible with your systems. You shall carry out virus/malware scanning on
>> your own before opening any attachment to this e-mail. The sender of this
>> e-mail and Company including its sister concerns shall not be liable for
>> any damage that may incur to you as a result of viruses, incompleteness of
>> this message, a delay in receipt of this message or any other computer
>> problems.
>> -------------- next part --------------
>> An HTML attachment was scrubbed...
>> URL: <http://lists.opensips.org/pipermail/users/attachments/20220308/d383bf19/attachment-0001.html>
>>
>> ------------------------------
>>
>> Message: 2
>> Date: Tue, 08 Mar 2022 11:51:45 +0100
>> From: Alain Bieuzent <alain.bieuzent at free.fr>
>> To: OpenSIPS users mailling list <users at lists.opensips.org>
>> Subject: Re: [OpenSIPS-Users] Header manipulation
>> Message-ID: <FA2BF751-1E38-4780-AC7D-682CFE3384B4 at free.fr>
>> Content-Type: text/plain; charset="utf-8"
>>
>> Hi Vinayak,
>>
>>
>>
>> Try with uac_replace_from([display],uri)
>>
>>
>>
>> Ragards
>>
>>
>>
>> De : Users <users-bounces at lists.opensips.org> au nom de Vinayak Makwana <vinayak.makwana at ecosmob.com>
>> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
>> Date : mardi 8 mars 2022 à 11:43
>> À : <users at lists.opensips.org>
>> Objet : [OpenSIPS-Users] Header manipulation
>>
>>
>>
>> Hello All,
>>
>> I want to replace uri-display in the FROM header using the avp_subst function. So, can anyone suggest a solution ?
>>
>> Here's My scenario:
>> Main:->  From:"abc"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>> After Changes ->  From:"pqrs"<sip:username at x.x.x.x;transport=UDP>;tag=6a8eda3f
>>
>> Many Thanks
>> Vinayak Makwana
>>
>>
>>
>> Disclaimer
>>
>> In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.
>>
>>
>>
>> Caution for viruses, malware etc.
>>
>> This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.
>>
>> _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> -------------- next part --------------
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>>
>> ------------------------------
>>
>> Subject: Digest Footer
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>> ------------------------------
>>
>> End of Users Digest, Vol 164, Issue 6
>> *************************************
>
> Disclaimer
> In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract 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 the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error.
>
> Caution for viruses, malware etc.
> This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.
> _______________________________________________ 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
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http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From ialex.gu at gmail.com  Fri Mar 11 17:59:04 2022
From: ialex.gu at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdC10Lkg0J3QuNC30LjQtdC90LrQvg==?=)
Date: Fri, 11 Mar 2022 20:59:04 +0300
Subject: [OpenSIPS-Users] segfault on nathelper.so
Message-ID: <CANtQs+UfT=M+WUMc_S0D6vDnWnRano1SHm7-5hcq19uYEONYeQ@mail.gmail.com>

Good day!
I have a trouble in opensips 3.2.5

version: opensips 3.2.5 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
main.c compiled on  with gcc 8

uname -a
Linux wsip 4.19.0-18-amd64 #1 SMP Debian 4.19.208-1 (2021-09-29)
x86_64 GNU/Linux

lsb_release -a
No LSB modules are available.
Distributor ID: Debian
Description:    Debian GNU/Linux 10 (buster)
Release:        10
Codename:       buster


[New LWP 7934]
[Thread debugging using libthread_db enabled]
Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
Core was generated by `/usr/sbin/opensips -P
/run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'.
Program terminated with signal SIGSEGV, Segmentation fault.
#0  0x00007f0937d67679 in remove_from_hash
(cell=cell at entry=0x7f0937f11e70) at nh_table.c:171
171     nh_table.c: No such file or directory


From y.kirsanov at gmail.com  Tue Mar 15 11:45:07 2022
From: y.kirsanov at gmail.com (Yury Kirsanov)
Date: Tue, 15 Mar 2022 22:45:07 +1100
Subject: [OpenSIPS-Users] tlt_mgm module - any way to pass cert/key as
	parameter for outgoing connection?
Message-ID: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>

Hi,
I've got a question, is there any way to pass SSL certificate and key as a
parameter to the tls_mgm module during script execution? For example, first
I do a REST request to our REST API server which returns me all required
parameters including certificate and key. Then I'd like to use this
response as a client certificate for outgoing connection to some
TLS-enabled server. Is there any way to do that? I know I can use DB module
and select a client certificate using avp variable, but that's not
convenient as it requires tls_reload MI command each time the DB is updated.

Thanks and best regards,
Yury.
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From venefax at gmail.com  Tue Mar 15 14:13:25 2022
From: venefax at gmail.com (Saint Michael)
Date: Tue, 15 Mar 2022 10:13:25 -0400
Subject: [OpenSIPS-Users] CDR not generated on 302 redirect
Message-ID: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>

My new business is to provide 302 Redirect services and Opensips does not
genrate a CDR for those calls. Other type of calls do generate a record. Is
this by design or is it a bug?
Every call that goes through Opensips should generate a record.
Any idea about what is going on?
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From abalashov at evaristesys.com  Tue Mar 15 14:35:00 2022
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 15 Mar 2022 10:35:00 -0400
Subject: [OpenSIPS-Users] CDR not generated on 302 redirect
In-Reply-To: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
References: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
Message-ID: <2A1B5BD6-B20E-40A7-947F-3EDD7788C7CB@evaristesys.com>


> On Mar 15, 2022, at 10:13 AM, Saint Michael <venefax at gmail.com> wrote:
> 
> Every call that goes through Opensips should generate a record. 

Is this just your opinion, or…?

-- 
Alex Balashov | Principal | Evariste Systems LLC

Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/



From david.villasmil.work at gmail.com  Tue Mar 15 14:40:32 2022
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Tue, 15 Mar 2022 14:40:32 +0000
Subject: [OpenSIPS-Users] CDR not generated on 302 redirect
In-Reply-To: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
References: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
Message-ID: <CAFGRPVoN8RMLfDRiJBoMWnZxUXhKkMs4jPwon7kyMFF9ExxuuQ@mail.gmail.com>

Look very carefully at the config. There’s probably somewhere it’s enabled.

Maybe  this
https://github.com/OpenSIPS/opensips/blob/master/examples/acc.cfg might
give you some ideas…

On Tue, 15 Mar 2022 at 14:14, Saint Michael <venefax at gmail.com> wrote:

> My new business is to provide 302 Redirect services and Opensips does not
> genrate a CDR for those calls. Other type of calls do generate a record. Is
> this by design or is it a bug?
> Every call that goes through Opensips should generate a record.
> Any idea about what is going on?
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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From Ben.Newlin at genesys.com  Tue Mar 15 14:43:47 2022
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Tue, 15 Mar 2022 14:43:47 +0000
Subject: [OpenSIPS-Users] CDR not generated on 302 redirect
In-Reply-To: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
References: <CAC9cSOC60EB07Uh2FR6hov9bZHT6qar_394ij3JUfQZM_z9cEQ@mail.gmail.com>
Message-ID: <BN0PR10MB5208835423DAB2198772ACDC95109@BN0PR10MB5208.namprd10.prod.outlook.com>

CDRs are generated for calls based on the dialogs. A call receiving a 302 Redirect does not establish a dialog so there will be no CDR.

OpenSIPS will generate an accounting record for any transaction if you want it to. Please review the documentation of the ACC module [1].

“failed - flag which indicates if the transaction should also be accounted in case of failure (status>=300);”

[1] - https://opensips.org/docs/modules/3.2.x/acc.html#func_do_accounting

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Saint Michael <venefax at gmail.com>
Date: Tuesday, March 15, 2022 at 10:15 AM
To: OpenSIPS users mailling list <Users at lists.opensips.org>
Subject: [OpenSIPS-Users] CDR not generated on 302 redirect
EXTERNAL EMAIL - Please use caution with links and attachments

________________________________
My new business is to provide 302 Redirect services and Opensips does not genrate a CDR for those calls. Other type of calls do generate a record. Is this by design or is it a bug?
Every call that goes through Opensips should generate a record.
Any idea about what is going on?
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From bogdan at opensips.org  Thu Mar 17 15:13:06 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 17 Mar 2022 17:13:06 +0200
Subject: [OpenSIPS-Users] =?utf-8?b?W0Jsb2ddIE9wZW5TSVBTIDMuMyDigJMgU3Rh?=
 =?utf-8?q?tus/readiness_and_reports_framework_in_OpenSIPS_3=2E3?=
Message-ID: <09425725-237a-477f-fc07-dab6e6748557@opensips.org>


A goof part of a software's value is given by how manageable it is under 
operational conditions. And in the context of the heavy migration to 
virtual machines and container based environments (like Kubernetes), 
“manageable” means to be able automatically spin up, terminate or 
monitor new instances, on demand, with zero human intervention.

So, *OpenSIPS 3.3* evolves, together with the world around it. And to 
address this automatic “manageability”, the 3.3 is bringing the 
*Status-Report framework*.

https://blog.opensips.org/2022/03/17/status-readiness-and-reports-framework-in-opensips-3-3/

Enjoy,

-- 
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp
   https://www.opensips.org/Training/Bootcamp

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From m.shirazi at gmail.com  Mon Mar 21 15:47:36 2022
From: m.shirazi at gmail.com (Mehdi Shirazi)
Date: Mon, 21 Mar 2022 19:17:36 +0330
Subject: [OpenSIPS-Users] SQL Cacher full caching auto reload specific key
Message-ID: <CAG2rjptEury8W1Z44kNrWpBFuJAm6fGg3jpNMF_X3crimDQ9fg@mail.gmail.com>

Hi
I use SQL Cacher in full caching mode. when the database changes I want to
automatically reload that specific key.
What is your suggestions for this?
With Mariadb trigger I cannot use system commands to reload that specific
key. Only way is using pooling method of changed records and use
sql_cacher_reload
?

Regards
Shirazi
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From andreigrav at gmail.com  Mon Mar 21 18:23:31 2022
From: andreigrav at gmail.com (Andrei G.)
Date: Mon, 21 Mar 2022 20:23:31 +0200
Subject: [OpenSIPS-Users]  registration status using mid_registrar
Message-ID: <CALUe069YPqoynrseM1+Uq_Pqt=5xhAK5k6USEGQZrBMa5yLaXg@mail.gmail.com>

Hi,

I'm trying to implement  Opensips with mid_registrar module in front of
Asterisk
from manual:

if (is_method("REGISTER")) {
	mid_registrar_save("location");
	switch ($retcode) {
	case 1:
		xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
		$ru = "sip:10.0.0.3:5070";
		if (!t_relay()) {
			send_reply(500, "Server Internal Error 1");
		}

		break;
	case 2:
		xlog("L_INFO", "REGISTER has been absorbed!\n");
		break;
	default:
		xlog("L_ERR", "mid-registrar error!\n");
		send_reply(500, "Server Internal Error 2");
	}

	exit;
}

Is there any way to check the status for REGISTER messages from main
registrar?

thank you
Andrei
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From kurgan-rus at inbox.ru  Mon Mar 21 18:40:44 2022
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Mon, 21 Mar 2022 21:40:44 +0300
Subject: [OpenSIPS-Users]
 =?utf-8?q?registration_status_using_mid=5Fregist?= =?utf-8?q?rar?=
In-Reply-To: <CALUe069YPqoynrseM1+Uq_Pqt=5xhAK5k6USEGQZrBMa5yLaXg@mail.gmail.com>
References: <CALUe069YPqoynrseM1+Uq_Pqt=5xhAK5k6USEGQZrBMa5yLaXg@mail.gmail.com>
Message-ID: <1647888044.411360993@f507.i.mail.ru>


Hi,
 
what if to try to use some custom t_on_reply route ?
Something like this:
 
 
if (is__method("REGISTER")) {
    xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
    $ru = "sip: 10.0.0.3:5070 ";
    t_on_reply("main_reg_replies");
 
    if (!t_relay()) {
        send_reply(500, "Server Internal Error 1");
    }
 
    …
}
 
…
 
onreply_route[main_reg_replies] {
    xlog("reply status: $rs\n");
}
 
-----------------------------------------------
BR, Alexey
https://alexeyka.zantsev.com/
 
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From andreigrav at gmail.com  Mon Mar 21 19:03:35 2022
From: andreigrav at gmail.com (Andrei G.)
Date: Mon, 21 Mar 2022 21:03:35 +0200
Subject: [OpenSIPS-Users] registration status using mid_registrar
In-Reply-To: <1647888044.411360993@f507.i.mail.ru>
References: <CALUe069YPqoynrseM1+Uq_Pqt=5xhAK5k6USEGQZrBMa5yLaXg@mail.gmail.com>
 <1647888044.411360993@f507.i.mail.ru>
Message-ID: <CALUe068Bvs5VtKjH8_Mt4Lwdwbn9sTH0MbUQ0LBEg00ianQKVQ@mail.gmail.com>

Perfect

Thank you
Andrei

On Mon, Mar 21, 2022 at 8:43 PM Alexey Kazantsev via Users <
users at lists.opensips.org> wrote:

> Hi,
>
> what if to try to use some custom t_on_reply route ?
> Something like this:
>
>
> if (is__method("REGISTER")) {
>     xlog("L_INFO", "forwarding REGISTER to main registrar...\n");
>     $ru = "sip:10.0.0.3:5070";
>     t_on_reply("main_reg_replies");
>
>     if (!t_relay()) {
>         send_reply(500, "Server Internal Error 1");
>     }
>
>     …
> }
>
> …
>
> onreply_route[main_reg_replies] {
>     xlog("reply status: $rs\n");
> }
>
> -----------------------------------------------
> BR, Alexey
> https://alexeyka.zantsev.com/
>
> _______________________________________________
> 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  Mon Mar 21 23:41:58 2022
From: alberto.rinaudo at gmail.com (Alberto)
Date: Mon, 21 Mar 2022 23:41:58 +0000
Subject: [OpenSIPS-Users] Python functions
Message-ID: <CANoA6_L1s5pkWRJhnDXHBu6CzS_ouXMUSWK7UsOBYYfkDyYF-Q@mail.gmail.com>

Hi lads and ladies,
I'm working on a python script called via python_exec, but I can't see any
function to do debug logs, except LM_ERR.

I tried
msg.call_function('log', str("test"))
or
msg.call_function('xlog', str("test"))

but I always get this error:
ERROR:python:opensips_LM_ERR: 37, SystemError, <method 'call_function' of
'OpenSIPS.msg' objects> returned a result with an error set

How should this be done?

Thanks
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From Johan at democon.be  Tue Mar 22 11:34:43 2022
From: Johan at democon.be (Johan De Clercq)
Date: Tue, 22 Mar 2022 12:34:43 +0100
Subject: [OpenSIPS-Users] inject_dtmf
Message-ID: <CAEVWGm-JeCKGOVOv4SCfHzM2okHWBSXgRevr6jyVcE6tpUS6Hg@mail.gmail.com>

Hi,

for one reason or another I don't get this working.
What I do
1. when the invite is send, i call rtpengine_offer with inject_DTMF flag.
2. in the onreply route, I call rtpengine_answer with inject_DTMF.  Then I
call rtpengine_playdtmf("0").

The dtmf is NEVER send out.

What do I do wrong here ?
Is there somebody with experience on this ?

wkr,
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From razvan at opensips.org  Wed Mar 23 08:17:42 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 23 Mar 2022 10:17:42 +0200
Subject: [OpenSIPS-Users] inject_dtmf
In-Reply-To: <CAEVWGm-JeCKGOVOv4SCfHzM2okHWBSXgRevr6jyVcE6tpUS6Hg@mail.gmail.com>
References: <CAEVWGm-JeCKGOVOv4SCfHzM2okHWBSXgRevr6jyVcE6tpUS6Hg@mail.gmail.com>
Message-ID: <0887ceec-eb99-ec5e-4a01-9baea87ab8c3@opensips.org>

Hi, Johan!

Can you post opensips logs of rtpengine module? Are there any errors.
Also, what version of OpenSIPS are you using?

Best regards,

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

On 3/22/22 13:34, Johan De Clercq wrote:
> Hi,
> 
> for one reason or another I don't get this working.
> What I do
> 1. when the invite is send, i call rtpengine_offer with inject_DTMF flag.
> 2. in the onreply route, I call rtpengine_answer with inject_DTMF.  Then 
> I call rtpengine_playdtmf("0").
> 
> The dtmf is NEVER send out.
> 
> What do I do wrong here ?
> Is there somebody with experience on this ?
> 
> wkr,
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From razvan at opensips.org  Wed Mar 23 08:20:38 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 23 Mar 2022 10:20:38 +0200
Subject: [OpenSIPS-Users] Python functions
In-Reply-To: <CANoA6_L1s5pkWRJhnDXHBu6CzS_ouXMUSWK7UsOBYYfkDyYF-Q@mail.gmail.com>
References: <CANoA6_L1s5pkWRJhnDXHBu6CzS_ouXMUSWK7UsOBYYfkDyYF-Q@mail.gmail.com>
Message-ID: <2addc0d6-0ca1-15fb-6ef6-526414e8ed73@opensips.org>

What OpenSIPS version are you using?

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

On 3/22/22 01:41, Alberto wrote:
> Hi lads and ladies,
> I'm working on a python script called via python_exec, but I can't see 
> any function to do debug logs, except LM_ERR.
> 
> I tried
> msg.call_function('log', str("test"))
> or
> msg.call_function('xlog', str("test"))
> 
> but I always get this error:
> ERROR:python:opensips_LM_ERR: 37, SystemError, <method 'call_function' 
> of 'OpenSIPS.msg' objects> returned a result with an error set
> 
> How should this be done?
> 
> Thanks
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From razvan at opensips.org  Wed Mar 23 08:22:16 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 23 Mar 2022 10:22:16 +0200
Subject: [OpenSIPS-Users] SQL Cacher full caching auto reload specific
	key
In-Reply-To: <CAG2rjptEury8W1Z44kNrWpBFuJAm6fGg3jpNMF_X3crimDQ9fg@mail.gmail.com>
References: <CAG2rjptEury8W1Z44kNrWpBFuJAm6fGg3jpNMF_X3crimDQ9fg@mail.gmail.com>
Message-ID: <214a06fb-0113-a9cf-7585-a46867decbcf@opensips.org>

Your assumption is correct - for full caching mode, only the entire 
table can be reloaded.
If you want to reload per record, you should be using on demand caching.

Best regards,

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

On 3/21/22 17:47, Mehdi Shirazi wrote:
> Hi
> I use SQL Cacher in full caching mode. when the database changes I want 
> to automatically reload that specific key.
> What is your suggestions for this?
> With Mariadb trigger I cannot use system commands to reload that 
> specific key. Only way is using pooling method of changed records and 
> use sql_cacher_reload ?
> 
> Regards
> Shirazi
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From Johan at democon.be  Wed Mar 23 08:23:53 2022
From: Johan at democon.be (Johan De Clercq)
Date: Wed, 23 Mar 2022 09:23:53 +0100
Subject: [OpenSIPS-Users] inject_dtmf
In-Reply-To: <0887ceec-eb99-ec5e-4a01-9baea87ab8c3@opensips.org>
References: <CAEVWGm-JeCKGOVOv4SCfHzM2okHWBSXgRevr6jyVcE6tpUS6Hg@mail.gmail.com>
 <0887ceec-eb99-ec5e-4a01-9baea87ab8c3@opensips.org>
Message-ID: <CAEVWGm845a7egcnbZDc0UtT9KhjMEHQPX0DRq3A_qB=dahx6Zw@mail.gmail.com>

I spoke with sipwise yesterday and according to them, the problem is that I
send the dtmf too early (immediately in reply route after
rtpengine_answer).
I will try this evening with setting up a call and then injecting the dtmf
with curl or something.

This leads me to the follow up question : how can I send a command from the
script after a certain time ?   I thought about timer-route, but there you
don't have the call id.

Op wo 23 mrt. 2022 om 09:20 schreef Răzvan Crainea <razvan at opensips.org>:

> Hi, Johan!
>
> Can you post opensips logs of rtpengine module? Are there any errors.
> Also, what version of OpenSIPS are you using?
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 3/22/22 13:34, Johan De Clercq wrote:
> > Hi,
> >
> > for one reason or another I don't get this working.
> > What I do
> > 1. when the invite is send, i call rtpengine_offer with inject_DTMF flag.
> > 2. in the onreply route, I call rtpengine_answer with inject_DTMF.  Then
> > I call rtpengine_playdtmf("0").
> >
> > The dtmf is NEVER send out.
> >
> > What do I do wrong here ?
> > Is there somebody with experience on this ?
> >
> > wkr,
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From alberto.rinaudo at gmail.com  Wed Mar 23 08:29:43 2022
From: alberto.rinaudo at gmail.com (Alberto)
Date: Wed, 23 Mar 2022 08:29:43 +0000
Subject: [OpenSIPS-Users] Python functions
In-Reply-To: <2addc0d6-0ca1-15fb-6ef6-526414e8ed73@opensips.org>
References: <CANoA6_L1s5pkWRJhnDXHBu6CzS_ouXMUSWK7UsOBYYfkDyYF-Q@mail.gmail.com>
 <2addc0d6-0ca1-15fb-6ef6-526414e8ed73@opensips.org>
Message-ID: <CANoA6_JtWNRH95LyH8m2ct8MVz-gZbqcF6u=YdZM+hdYbmrpKA@mail.gmail.com>

Hi,
3.2
Sorry for not mentioning it earlier

On Wed, 23 Mar 2022, 08:22 Răzvan Crainea, <razvan at opensips.org> wrote:

> What OpenSIPS version are you using?
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 3/22/22 01:41, Alberto wrote:
> > Hi lads and ladies,
> > I'm working on a python script called via python_exec, but I can't see
> > any function to do debug logs, except LM_ERR.
> >
> > I tried
> > msg.call_function('log', str("test"))
> > or
> > msg.call_function('xlog', str("test"))
> >
> > but I always get this error:
> > ERROR:python:opensips_LM_ERR: 37, SystemError, <method 'call_function'
> > of 'OpenSIPS.msg' objects> returned a result with an error set
> >
> > How should this be done?
> >
> > Thanks
> >
> > _______________________________________________
> > 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 spanda at 3clogic.com  Wed Mar 23 11:40:44 2022
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 23 Mar 2022 17:10:44 +0530
Subject: [OpenSIPS-Users] rtp_relay module implementation help .
Message-ID: <CALf-Q46oykHEaMK2bjBMcOUC7si_ZYo15aGa39YmLB7NU5FzpQ@mail.gmail.com>

Hi All ,

I was going through the doc and did a simple POC on rtp_relay module and
the media server is rtpengine .

My config file looks like below .

loadmodule "dialog.so"
loadmodule "rtp_relay.so"
loadmodule "rtpengine.so"
modparam("rtpengine", "rtpengine_sock", "udp:20.0.x.x:22000=3")
modparam("rtpengine", "rtpengine_sock", "udp:20.0.x.y:22000=0")

route{
.....
             if (is_method("INVITE")){
                             $rtp_relay = "replace-origin
replace-session-connection";
                        $rtp_relay_peer = "replace-origin
replace-session-connection";
                        #rtp_relay_engage("rtpproxy");
                        rtp_relay_engage("rtpengine");
......................
   }
 }

While running this if rtpengine becomes unreachable through which media
session is established , then opensips automatically wont switch the same
call to another rtpengine node .  I have to run opensips-cli command to
switch the rtpengine .

/usr/local/bin/opensips-cli -x mi rtp_relay_update engine=rtpengine set=0
node=udp:20.0.x.x:22000 new_node=udp:20.0.x.y:22000

Automatic switching possible or not? If possible then how ? What should I
do for the automatic switching of rtpengine nodes ? Media high availability
is only possible  if opensips will automatically switch the defective rtp
node to the running one .

Please do suggest .

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From sbirla at tampahost.net  Wed Mar 23 14:00:00 2022
From: sbirla at tampahost.net (Sumit Birla)
Date: Wed, 23 Mar 2022 10:00:00 -0400
Subject: [OpenSIPS-Users] Upgrade OpenSIPS version 2.3 to 3.2
Message-ID: <A88E1B03-B91B-436E-A8E4-6D1CC3350407@tampahost.net>

Hi all,

A couple of questions about upgrading an old instance on OpenSIPS:

Is it possible to update OpenSIPS database from version 2.3 to 3.2 in one shot?   Or do I have to go through the steps:

2.3 -> 2.4 -> 3.0 -> 3.1 -> 3.2

If I install version 3.2,  will it have the capability to migrate the database through the various versions,  or do I need to install corresponding versions of OpenSIPS?

Thanks.

From razvan at opensips.org  Wed Mar 23 14:09:34 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 23 Mar 2022 16:09:34 +0200
Subject: [OpenSIPS-Users] Upgrade OpenSIPS version 2.3 to 3.2
In-Reply-To: <A88E1B03-B91B-436E-A8E4-6D1CC3350407@tampahost.net>
References: <A88E1B03-B91B-436E-A8E4-6D1CC3350407@tampahost.net>
Message-ID: <e36d2e42-c678-d84f-2007-05aab087c8ed@opensips.org>

Hi, Sumit!

You need to gradually migrate the DB from 2.3 to 2.4, then 3.0, etc. You 
don't need to install opensips for that, all you need is the database 
schema.

Best regards,

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

On 3/23/22 16:00, Sumit Birla wrote:
> Hi all,
> 
> A couple of questions about upgrading an old instance on OpenSIPS:
> 
> Is it possible to update OpenSIPS database from version 2.3 to 3.2 in one shot?   Or do I have to go through the steps:
> 
> 2.3 -> 2.4 -> 3.0 -> 3.1 -> 3.2
> 
> If I install version 3.2,  will it have the capability to migrate the database through the various versions,  or do I need to install corresponding versions of OpenSIPS?
> 
> Thanks.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From razvan at opensips.org  Wed Mar 23 14:13:55 2022
From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=)
Date: Wed, 23 Mar 2022 16:13:55 +0200
Subject: [OpenSIPS-Users] rtp_relay module implementation help .
In-Reply-To: <CALf-Q46oykHEaMK2bjBMcOUC7si_ZYo15aGa39YmLB7NU5FzpQ@mail.gmail.com>
References: <CALf-Q46oykHEaMK2bjBMcOUC7si_ZYo15aGa39YmLB7NU5FzpQ@mail.gmail.com>
Message-ID: <60aa2245-ccd9-67ba-40a2-f46a8372dab9@opensips.org>

Hi, Sasmita!

There is no auto-switching mode, you will have to do it manually.
You need to monitor rtpengine through external scripts, and when it 
breaks, run the opensips-cli rtp_relay_update command.

Best regards,

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

On 3/23/22 13:40, Sasmita Panda wrote:
> Hi All ,
> 
> I was going through the doc and did a simple POC on rtp_relay module and 
> the media server is rtpengine .
> 
> My config file looks like below .
> 
> loadmodule "dialog.so"
> loadmodule "rtp_relay.so"
> loadmodule "rtpengine.so"
> modparam("rtpengine", "rtpengine_sock", "udp:20.0.x.x:22000=3")
> modparam("rtpengine", "rtpengine_sock", "udp:20.0.x.y:22000=0")
> 
> route{
> .....
>               if (is_method("INVITE")){
>                               $rtp_relay = "replace-origin 
> replace-session-connection";
>                          $rtp_relay_peer = "replace-origin 
> replace-session-connection";
>                          #rtp_relay_engage("rtpproxy");
>                          rtp_relay_engage("rtpengine");
> ......................
>     }
>   }
> 
> While running this if rtpengine becomes unreachable through which media 
> session is established , then opensips automatically wont switch the 
> same call to another rtpengine node .  I have to run opensips-cli 
> command to switch the rtpengine .
> 
> /usr/local/bin/opensips-cli -x mi rtp_relay_update engine=rtpengine 
> set=0 node=udp:20.0.x.x:22000 new_node=udp:20.0.x.y:22000
> 
> Automatic switching possible or not? If possible then how ? What should 
> I do for the automatic switching of rtpengine nodes ? Media high 
> availability is only possible  if opensips will automatically switch the 
> defective rtp node to the running one .
> 
> Please do suggest .
> 
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From andreigrav at gmail.com  Wed Mar 23 16:07:57 2022
From: andreigrav at gmail.com (Andrei G.)
Date: Wed, 23 Mar 2022 18:07:57 +0200
Subject: [OpenSIPS-Users]  mid_registrar multi domains
Message-ID: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>

Hey guys,

I successfully tested opensips with mid_registrar for one domain

Is it possible to use opensips in front of 2 asterisk boxes and redirect
registrations based on a prefix username, not domain prefix?

Something like
asterisk1-user at mid-registrar.domain - where opensips manage registration
for user at asterisk1
asterisk2-user at mid-registrar.domain - where opensips manage registration
for user at asterisk2

Regards
Andrei G.
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From stefan.carlsson at tetab.nu  Wed Mar 23 18:59:34 2022
From: stefan.carlsson at tetab.nu (Stefan Carlsson)
Date: Wed, 23 Mar 2022 18:59:34 +0000
Subject: [OpenSIPS-Users] mysql_db  installation fails ...
Message-ID: <ff42427099ab42e5854468dcc08a1dc8@tetab.nu>

Hi !

A fresh install of opensips 3.2.5 (x86_64/linux) , now I try to install mysql_db module, but fails.

Tried to install via yum,  got this ....

yum install opensips-mysql
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: ftpmirror.infania.net
* centos-sclo-rh: ftpmirror.infania.net
* centos-sclo-sclo: ftpmirror.infania.net
* epel: mirrors.glesys.net
* extras: ftpmirror.infania.net
* remi-php74: mirror.netsite.dk
* remi-safe: mirror.netsite.dk
* updates: ftpmirror.infania.net
Resolving Dependencies
--> Running transaction check
---> Package opensips-mysql.x86_64 0:1.10.5-4.el7 will be installed
--> Processing Dependency: opensips(x86-64) = 1.10.5-4.el7 for package: opensips-mysql-1.10.5-4.el7.x86_64
--> Finished Dependency Resolution
Error: Package: opensips-mysql-1.10.5-4.el7.x86_64 (epel)
           Requires: opensips(x86-64) = 1.10.5-4.el7
           Installed: opensips-3.2.5-1.el7.x86_64 (@opensips)
               opensips(x86-64) = 3.2.5-1.el7
           Available: opensips-1.10.5-4.el7.x86_64 (epel)
               opensips(x86-64) = 1.10.5-4.el7
           Available: opensips-3.2.0-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.0-1.el7
           Available: opensips-3.2.2-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.2-1.el7
           Available: opensips-3.2.3-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.3-1.el7
           Available: opensips-3.2.4-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.4-1.el7
You could try using --skip-broken to work around the problem
You could try running: rpm -Va --nofiles --nodigest

Any ideas ....


Regards / Mvh  Stefan





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From osas at voipembedded.com  Wed Mar 23 22:54:57 2022
From: osas at voipembedded.com (Ovidiu Sas)
Date: Wed, 23 Mar 2022 18:54:57 -0400
Subject: [OpenSIPS-Users] OpenSIPS timers
Message-ID: <CAND0Lkvi=YXr_puy3ewEBWhUHfpMJADqZB+oyMPd=3SUVQeaDw@mail.gmail.com>

Hello all,

I'm working on tuning an opensips server. I get this pesky:
WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled
I was trying to get rid of them by playing with the tm
timer_partitions parameter and the timer_workers core param.
By increasing any of them doesn't increase performance.
By increasing both of them, it actually decreases performance.
The server is not at limit, the load on the UDP workers is around
50-60 with some spikes.
I have around 3500+ cps sipp traffic.

My understanding is that by increasing the number of timer_partitions,
we will have more procs walking in parallel over the timer structures.
If we have on timer structure, we have one proc walking over it.
How is this working for two timer structures? What is the difference
between the first and the second timer structure? Should we expect
less work for each proc?

For now, to reduce the occurrence of the warning log, I increased the
timer interval for tm-utimer from 100ms to 200ms. This should be ok as
the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set.

Thanks,
Ovidiu

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


From alberto.rinaudo at gmail.com  Wed Mar 23 23:31:34 2022
From: alberto.rinaudo at gmail.com (Alberto)
Date: Wed, 23 Mar 2022 23:31:34 +0000
Subject: [OpenSIPS-Users] opensips 3.2 and latest github rtpengine
Message-ID: <CANoA6_K172OsLYbAurRN6+boQ9vAwSMRrWkDxGxEsMtPuMH4-w@mail.gmail.com>

Hi,

I'm trying to change the session name, the s= line, while using rtpengine.

If I remove rtpengine and do replace_body_all("^s=.*$", "s=abczzz"); it
works just fine and I see the new session name in the second leg of the
call.

But when rtpengine_offer is called, the original sdp body is used instead
of the modified body.
I tried to do replace_body_all before and after rtpengine_offer, but it
doesn't work, the second leg always has the original session name.

Any advice?
Thanks
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From stefan.carlsson at tetab.nu  Wed Mar 23 22:53:12 2022
From: stefan.carlsson at tetab.nu (Stefan Carlsson)
Date: Wed, 23 Mar 2022 22:53:12 +0000
Subject: [OpenSIPS-Users] mysql_db  installation fails ...
In-Reply-To: <ff42427099ab42e5854468dcc08a1dc8@tetab.nu>
References: <ff42427099ab42e5854468dcc08a1dc8@tetab.nu>
Message-ID: <7c3336f4f1544d41b16c9369e8b06966@tetab.nu>

Found the "problem"

Forgot the -module in modulename ...

Problem solved...

From: Users <users-bounces at lists.opensips.org> On Behalf Of Stefan Carlsson
Sent: Wednesday, 23 March, 2022 20:00
To: users at lists.opensips.org
Subject: [OpenSIPS-Users] mysql_db installation fails ...

Hi !

A fresh install of opensips 3.2.5 (x86_64/linux) , now I try to install mysql_db module, but fails.

Tried to install via yum,  got this ....

yum install opensips-mysql
Loaded plugins: fastestmirror
Loading mirror speeds from cached hostfile
* base: ftpmirror.infania.net
* centos-sclo-rh: ftpmirror.infania.net
* centos-sclo-sclo: ftpmirror.infania.net
* epel: mirrors.glesys.net
* extras: ftpmirror.infania.net
* remi-php74: mirror.netsite.dk
* remi-safe: mirror.netsite.dk
* updates: ftpmirror.infania.net
Resolving Dependencies
--> Running transaction check
---> Package opensips-mysql.x86_64 0:1.10.5-4.el7 will be installed
--> Processing Dependency: opensips(x86-64) = 1.10.5-4.el7 for package: opensips-mysql-1.10.5-4.el7.x86_64
--> Finished Dependency Resolution
Error: Package: opensips-mysql-1.10.5-4.el7.x86_64 (epel)
           Requires: opensips(x86-64) = 1.10.5-4.el7
           Installed: opensips-3.2.5-1.el7.x86_64 (@opensips)
               opensips(x86-64) = 3.2.5-1.el7
           Available: opensips-1.10.5-4.el7.x86_64 (epel)
               opensips(x86-64) = 1.10.5-4.el7
           Available: opensips-3.2.0-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.0-1.el7
           Available: opensips-3.2.2-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.2-1.el7
           Available: opensips-3.2.3-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.3-1.el7
           Available: opensips-3.2.4-1.el7.x86_64 (opensips)
               opensips(x86-64) = 3.2.4-1.el7
You could try using --skip-broken to work around the problem
You could try running: rpm -Va --nofiles --nodigest

Any ideas ....


Regards / Mvh  Stefan





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From stefan.carlsson at tetab.nu  Wed Mar 23 23:18:04 2022
From: stefan.carlsson at tetab.nu (Stefan Carlsson)
Date: Wed, 23 Mar 2022 23:18:04 +0000
Subject: [OpenSIPS-Users] Opensips 3 module names...
Message-ID: <9b5c7abc-64c0-4dee-9a81-733f8619a6a3@tetab.nu>

Hi ....

Where can I find the modules names so I can install the via yum.

Is it in the form:   opensips-[modulename]-module

Need the proto modules ....

It seems to be changed a bit after rel.2

Kind Regards

Stefan
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From artiom.druz at gmail.com  Thu Mar 24 02:17:38 2022
From: artiom.druz at gmail.com (Artiom Druz)
Date: Thu, 24 Mar 2022 07:17:38 +0500
Subject: [OpenSIPS-Users] mid_registrar multi domains
In-Reply-To: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>
References: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>
Message-ID: <CAEBOSMTxJGWaCCE_6BpdeVjG72hiGzm=J+cndMUVaCFaOONK8Q@mail.gmail.com>

Hello, Andrei.
You can do that by taking $tU (user from "To" header) and using some
transformation function. In your example can be used s.select function. (
https://www.opensips.org/Documentation/Script-Tran-3-2#toc7)

Best regards,
Artiom Druz

ср, 23 мар. 2022 г., 21:11 Andrei G. <andreigrav at gmail.com>:

> Hey guys,
>
> I successfully tested opensips with mid_registrar for one domain
>
> Is it possible to use opensips in front of 2 asterisk boxes and redirect
> registrations based on a prefix username, not domain prefix?
>
> Something like
> asterisk1-user at mid-registrar.domain - where opensips manage registration
> for user at asterisk1
> asterisk2-user at mid-registrar.domain - where opensips manage registration
> for user at asterisk2
>
> Regards
> Andrei G.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From artiom.druz at gmail.com  Thu Mar 24 02:30:48 2022
From: artiom.druz at gmail.com (Artiom Druz)
Date: Thu, 24 Mar 2022 07:30:48 +0500
Subject: [OpenSIPS-Users] opensips 3.2 and latest github rtpengine
In-Reply-To: <CANoA6_K172OsLYbAurRN6+boQ9vAwSMRrWkDxGxEsMtPuMH4-w@mail.gmail.com>
References: <CANoA6_K172OsLYbAurRN6+boQ9vAwSMRrWkDxGxEsMtPuMH4-w@mail.gmail.com>
Message-ID: <CAEBOSMRdc7C0ZpTu_e=1VcDaW7=q_p3xcGvb5dqrqNnyQP3hHA@mail.gmail.com>

Hello, Alberto.
You can modify it by using an optional parameter in rtpengine_offer
(sdp_var -
https://opensips.org/html/docs/modules/3.2.x/rtpengine#func_rtpengine_offer
).
Logic:
You can write new sdp body to the variable instead of rewrite of existing
SDP. After that you can modify "s" parameter in this variable.
Next - you delete existing SDP (remove_body_part()) and add new SDP with
content from variable (add_body_part()).

Best regards,
Artiom Druz

чт, 24 мар. 2022 г., 04:34 Alberto <alberto.rinaudo at gmail.com>:

> Hi,
>
> I'm trying to change the session name, the s= line, while using rtpengine.
>
> If I remove rtpengine and do replace_body_all("^s=.*$", "s=abczzz"); it
> works just fine and I see the new session name in the second leg of the
> call.
>
> But when rtpengine_offer is called, the original sdp body is used instead
> of the modified body.
> I tried to do replace_body_all before and after rtpengine_offer, but it
> doesn't work, the second leg always has the original session name.
>
> Any advice?
> Thanks
> _______________________________________________
> 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  Thu Mar 24 12:56:41 2022
From: alberto.rinaudo at gmail.com (Alberto)
Date: Thu, 24 Mar 2022 12:56:41 +0000
Subject: [OpenSIPS-Users] opensips 3.2 and latest github rtpengine
In-Reply-To: <CAEBOSMRdc7C0ZpTu_e=1VcDaW7=q_p3xcGvb5dqrqNnyQP3hHA@mail.gmail.com>
References: <CANoA6_K172OsLYbAurRN6+boQ9vAwSMRrWkDxGxEsMtPuMH4-w@mail.gmail.com>
 <CAEBOSMRdc7C0ZpTu_e=1VcDaW7=q_p3xcGvb5dqrqNnyQP3hHA@mail.gmail.com>
Message-ID: <CANoA6_LPpct-OMXJriRvEwSvbjoTMgB0cxasHDWA8ro+uELHrQ@mail.gmail.com>

That works, here's what I've ended up using:

rtpengine_manage("loop-protect ... other things I need", , $var(body));
remove_body_part();
add_body_part($(var(body){re.subst,/^s=.*$/s=abcxxx/g}), "application/sdp");

but seems cumbersome, I was expecting to do rtpengine_manage followed by
replace_body_all and don't interfere further with the flow

Anyway, thanks

On Thu, 24 Mar 2022 at 02:33, Artiom Druz <artiom.druz at gmail.com> wrote:

> Hello, Alberto.
> You can modify it by using an optional parameter in rtpengine_offer
> (sdp_var -
> https://opensips.org/html/docs/modules/3.2.x/rtpengine#func_rtpengine_offer
> ).
> Logic:
> You can write new sdp body to the variable instead of rewrite of existing
> SDP. After that you can modify "s" parameter in this variable.
> Next - you delete existing SDP (remove_body_part()) and add new SDP with
> content from variable (add_body_part()).
>
> Best regards,
> Artiom Druz
>
> чт, 24 мар. 2022 г., 04:34 Alberto <alberto.rinaudo at gmail.com>:
>
>> Hi,
>>
>> I'm trying to change the session name, the s= line, while using rtpengine.
>>
>> If I remove rtpengine and do replace_body_all("^s=.*$", "s=abczzz"); it
>> works just fine and I see the new session name in the second leg of the
>> call.
>>
>> But when rtpengine_offer is called, the original sdp body is used instead
>> of the modified body.
>> I tried to do replace_body_all before and after rtpengine_offer, but it
>> doesn't work, the second leg always has the original session name.
>>
>> Any advice?
>> Thanks
>> _______________________________________________
>> 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 kurgan-rus at inbox.ru  Thu Mar 24 19:29:25 2022
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Thu, 24 Mar 2022 22:29:25 +0300
Subject: [OpenSIPS-Users]
 =?utf-8?q?mysql=5Fdb__installation_fails_=2E=2E?= =?utf-8?q?=2E?=
In-Reply-To: <7c3336f4f1544d41b16c9369e8b06966@tetab.nu>
References: <ff42427099ab42e5854468dcc08a1dc8@tetab.nu>
 <7c3336f4f1544d41b16c9369e8b06966@tetab.nu>
Message-ID: <1648150165.29564018@f381.i.mail.ru>


Hello Stefan,
 
as we see, the packages are available from 2 repos in your system —
an old version from ‘epel’ and an up-to-date one from ‘opensips’.
 
It can be useful to run such command to list available packages
and repositories they belong to:
 
    yum --showduplicates list available opensips*
 
And then instruct yum exactly to install some certain version of the package, e.g.:
 
    yum install opensips-X.Y.Z-3.el7
 
It’s also possible to specify the repository from which you intend to install a package:
 
    yum --enablerepo=opensips install opensips
 
 
Maybe these notes will be useful for somebody, as you’ve already solved the problem.
 
-----------------------------------------------
BR, Alexey
https://alexeyka.zantsev.com/
 
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From kurgan-rus at inbox.ru  Thu Mar 24 19:39:06 2022
From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=)
Date: Thu, 24 Mar 2022 22:39:06 +0300
Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_3_module_names=2E=2E=2E?=
In-Reply-To: <9b5c7abc-64c0-4dee-9a81-733f8619a6a3@tetab.nu>
References: <9b5c7abc-64c0-4dee-9a81-733f8619a6a3@tetab.nu>
Message-ID: <1648150746.325564799@f381.i.mail.ru>


Hi Stefan,
 
as I understood, you use CentOS, so 
this command should be exactly what you need:
 
    yum --showduplicates list available opensips*
 
You may also browse repository:
https://yum.opensips.org/3.2/releases/el/7/x86_64/
https://yum.opensips.org/browse.php
-----------------------------------------------
BR, Alexey
https://alexeyka.zantsev.com/
 
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From marcin at voipplus.net  Fri Mar 25 15:35:00 2022
From: marcin at voipplus.net (Marcin Groszek)
Date: Fri, 25 Mar 2022 10:35:00 -0500
Subject: [OpenSIPS-Users] driver error(2026)
Message-ID: <0e0e9b06-8fae-a278-4bf4-a223e09d585c@voipplus.net>

opensips version 3.1.5 compiled on oracle linux 8

version: opensips 3.1.5 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: a21fbc16e

When |tls_mgm.so is used and ||modparam("db_mysql", "use_tls", 1)
mysql  Ver 15.1 Distrib 10.5.15-MariaDB, for Linux (x86_64) using 
readline 5.1|

|Opensips has issue starting with following errors:|

| ERROR:db_mysql:db_mysql_connect: driver error(2026): Unknown SSL error
  ERROR:db_mysql:db_mysql_connect: driver error(2026): SSL connection 
error: PEM lib|

|After 2-5 attempts opensips does start and communicates with database.
|

|I found the bug report on github at 
https://github.com/OpenSIPS/opensips/issues/2594 but it went without any 
respond.|

|Debug and configuration at the bug report match my results.|

|
|

||

-- 
Best Regards:
Marcin Groszek
Business Voip Resource.
http://www.voipplus.net

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From alexanderhenryperkins at gmail.com  Fri Mar 25 20:20:43 2022
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Fri, 25 Mar 2022 16:20:43 -0400
Subject: [OpenSIPS-Users] Requesting IP
Message-ID: <CALLkTp0-2+fB78yE5WrD289XtUuNWmEGzwDjtP2spY+bw2OY0g@mail.gmail.com>

Hi All.  I have a silly question - which variable should I use to get the
IP that is connecting to my OpenSIPs box?

Thanks all,
Alex
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From marcin at voipplus.net  Fri Mar 25 20:40:41 2022
From: marcin at voipplus.net (Marcin Groszek)
Date: Fri, 25 Mar 2022 15:40:41 -0500
Subject: [OpenSIPS-Users] Requesting IP
In-Reply-To: <CALLkTp0-2+fB78yE5WrD289XtUuNWmEGzwDjtP2spY+bw2OY0g@mail.gmail.com>
References: <CALLkTp0-2+fB78yE5WrD289XtUuNWmEGzwDjtP2spY+bw2OY0g@mail.gmail.com>
Message-ID: <20c25763-2b4e-60e3-15b7-e5a2b5393ecf@voipplus.net>

What you are looking for is :  $si

see https://www.opensips.org/Documentation/Script-CoreVar-3-1#toc84


On 3/25/2022 3:20 PM, Alexander Perkins wrote:
> Hi All.  I have a silly question - which variable should I use to get 
> the IP that is connecting to my OpenSIPs box?
>
> Thanks all,
> Alex
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Best Regards:
Marcin Groszek
Business Voip Resource.
http://www.voipplus.net

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From andreigrav at gmail.com  Mon Mar 28 07:55:06 2022
From: andreigrav at gmail.com (Andrei G.)
Date: Mon, 28 Mar 2022 10:55:06 +0300
Subject: [OpenSIPS-Users] mid_registrar multi domains
In-Reply-To: <CAEBOSMTxJGWaCCE_6BpdeVjG72hiGzm=J+cndMUVaCFaOONK8Q@mail.gmail.com>
References: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>
 <CAEBOSMTxJGWaCCE_6BpdeVjG72hiGzm=J+cndMUVaCFaOONK8Q@mail.gmail.com>
Message-ID: <CALUe069Kibr2R0ZyWkjJWDqgb7Z51vbDUh0DG12y466XFvLCHA@mail.gmail.com>

Hi,

How do I modify the field "Authorization: Digest username" in REGISTER
packet?

tried with uac_replace_to(,"user at asterisk1") and
uac_replace_from(,"user at asterisk1") but it does affect only to and from
headers not digest username

thanks
Andrei


On Thu, Mar 24, 2022 at 4:20 AM Artiom Druz <artiom.druz at gmail.com> wrote:

> Hello, Andrei.
> You can do that by taking $tU (user from "To" header) and using some
> transformation function. In your example can be used s.select function. (
> https://www.opensips.org/Documentation/Script-Tran-3-2#toc7)
>
> Best regards,
> Artiom Druz
>
> ср, 23 мар. 2022 г., 21:11 Andrei G. <andreigrav at gmail.com>:
>
>> Hey guys,
>>
>> I successfully tested opensips with mid_registrar for one domain
>>
>> Is it possible to use opensips in front of 2 asterisk boxes and redirect
>> registrations based on a prefix username, not domain prefix?
>>
>> Something like
>> asterisk1-user at mid-registrar.domain - where opensips manage registration
>> for user at asterisk1
>> asterisk2-user at mid-registrar.domain - where opensips manage registration
>> for user at asterisk2
>>
>> Regards
>> Andrei G.
>>
>> _______________________________________________
>> 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 chester at zigbang.com  Mon Mar 28 08:17:14 2022
From: chester at zigbang.com (=?UTF-8?B?7J206riw7JuQ?=)
Date: Mon, 28 Mar 2022 17:17:14 +0900
Subject: [OpenSIPS-Users] strange INVITE transmission
Message-ID: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>

Hi guys, I'm trying to introduce opensips into my company these days.

I'm testing with many other phones. During the test, I had a very strange
symptom.

Please, take look at the picture below first. (Please find my screenshot
from the link below)
A. The most left one 14.52.252.236 is a phone (with hardware).
B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
58.79.209.75
C. 175.223.34.31 is a zoiper softphone which is working on iphone
D. 192.168.10.187 is the private IP address of A - a phone which is
the most left one (14.52.252.236)



https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing

As you can see, C is calling A but opensips transmits the INVITE rqeuset to
A's private IP address.
INVITE request should be transmitted A's public IP address - In this
scenario 14.52.252.236 (A)

I also attach my opensips.cfg. What's wrong with me?

Thank you


Regards
Kiwon
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From daniel.zanutti at gmail.com  Mon Mar 28 13:45:22 2022
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Mon, 28 Mar 2022 10:45:22 -0300
Subject: [OpenSIPS-Users] strange INVITE transmission
In-Reply-To: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>
References: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>
Message-ID: <CAN0jgiNzT3xEKfNMkYLtAEbF9Z_EMxz7D+sWWLFXr_eBWoBx0Q@mail.gmail.com>

Hi Kiwon

You need to handle NAT scenarios. Try putting this code on line 254, right
after "t_check_trans()":

if (nat_uac_test("7"))
{
  #nathelper
  if(is_method("REGISTER"))
    fix_nated_register();
  else
    fix_nated_contact();
  xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
ID=$ci\n");
}

You also need to enable nathelper module. The reason is that you need to
use the public IP/Port that sent data to opensips and ignore the Contact.


On Mon, Mar 28, 2022 at 5:20 AM 이기원 <chester at zigbang.com> wrote:

> Hi guys, I'm trying to introduce opensips into my company these days.
>
> I'm testing with many other phones. During the test, I had a very strange
> symptom.
>
> Please, take look at the picture below first. (Please find my screenshot
> from the link below)
> A. The most left one 14.52.252.236 is a phone (with hardware).
> B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
> 58.79.209.75
> C. 175.223.34.31 is a zoiper softphone which is working on iphone
> D. 192.168.10.187 is the private IP address of A - a phone which is
> the most left one (14.52.252.236)
>
>
>
>
> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>
> As you can see, C is calling A but opensips transmits the INVITE rqeuset
> to A's private IP address.
> INVITE request should be transmitted A's public IP address - In this
> scenario 14.52.252.236 (A)
>
> I also attach my opensips.cfg. What's wrong with me?
>
> Thank you
>
>
> Regards
> Kiwon
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From andreigrav at gmail.com  Mon Mar 28 19:09:19 2022
From: andreigrav at gmail.com (Andrei G.)
Date: Mon, 28 Mar 2022 22:09:19 +0300
Subject: [OpenSIPS-Users] mid_registrar multi domains
In-Reply-To: <CALUe069Kibr2R0ZyWkjJWDqgb7Z51vbDUh0DG12y466XFvLCHA@mail.gmail.com>
References: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>
 <CAEBOSMTxJGWaCCE_6BpdeVjG72hiGzm=J+cndMUVaCFaOONK8Q@mail.gmail.com>
 <CALUe069Kibr2R0ZyWkjJWDqgb7Z51vbDUh0DG12y466XFvLCHA@mail.gmail.com>
Message-ID: <CALUe0694WLhzT6gXXFegfNVAxSx+2xvZs1mJW6pAFbPuMV3zCw@mail.gmail.com>

Hello,

I'm using the following code in opensips.config

#############################################################################################
        # absorb retransmissions, but do not create transaction
        t_check_trans();

        if (is_method("REGISTER")) {
            uac_replace_to(,"sip:user1 at asterisk1.local");
            uac_replace_from(,"sip:user1 at asterisk1.local");
            mid_registrar_save("location","m");
              switch ($retcode) {
                 case 1:
                    $ru = "sip:asterisk1.local";
                    t_on_reply("main_reg_replies");
                    t_relay();
                    break;
                 case 2:
                    #xlog("[LOG] absorbing REGISTER! ($$ci=$ci)\n");
                    break;
                 default:
                    xlog("[LOG] failed to save registration! ($$ci=$ci)\n");
             }
              exit;
        }

        if ( is_method("INVITE|MESSAGE|CANCEL|BYE|NOTIFY") && ($si !=
"IP_ASTERISK1" || $sp != 5060) ) {
            $ru = "sip:" + $tU + "@asterisk1.local";
        }
#############################################################################################


*U 2022/03/28 21:52:40.848274 IP_CLIENT:19606 -> IP_OPENSIPS:5060 #1*
REGISTER sip:OPENSIPS SIP/2.0.
Via: SIP/2.0/UDP 192.168.11.1:19606
;received=IP_CLIENT;rport=19606;branch=z9hG4bK-9p6796298349846258764r.
From: <sip:user1-asterisk at IP_OPENSIPS:5060>;tag=8g7970426010400195798m.
To: <sip:user1-asterisk at IP_OPENSIPS:5060>.
Call-ID: 6e6985069689257392584k25377rmwpR.
CSeq: 7319 REGISTER.
Max-Forwards: 70.
Authorization: Digest
username="user1-asterisk",realm="asterisk",nonce="3aa213ca",uri="sip:IP_OPENSIPS:5060",response="89d2d2d37a0a348960a64f9caf40b96a",algorithm=MD5.
Contact: <sip:user1-asterisk at 192.168.11.1:19606>;expires=0.
Supported: replaces.
Content-Length: 0.


*U 2022/03/28 21:52:40.853495 IP_OPENSIPS:5060 -> IP_ASTERISK1:5060 #2*
REGISTER sip:asterisk1.local:5060 SIP/2.0.
Via: SIP/2.0/UDP IP_OPENSIPS:5060;branch=z9hG4bKd2e8.d0acd904.0.
Via: SIP/2.0/UDP 192.168.11.1:19606
;received=IP_CLIENT;rport=19606;branch=z9hG4bK-9p6796298349846258764r.
From: <sip:user1 at asterisk1.local>;tag=8g7970426010400195798m.
To: <sip:user1 at asterisk1.local>.
Call-ID: 6e6985069689257392584k25377rmwpR.
CSeq: 7319 REGISTER.
Max-Forwards: 69.
Authorization: Digest username="user1-asterisk"
,realm="asterisk",nonce="3aa213ca",uri="sip:IP_OPENSIPS:5060",response="89d2d2d37a0a348960a64f9caf40b96a",algorithm=MD5.
Contact: <sip:user1-asterisk%40IP_OPENSIPS at IP_OPENSIPS:5060>;expires=0.
Supported: replaces.
Allow:
ACK,PRACK,BYE,CANCEL,INVITE,UPDATE,MESSAGE,INFO,OPTIONS,SUBSCRIBE,NOTIFY,REFER.
Content-Length: 0.

Please help me how to modify  Digest username="user1"

thanks
Andrei

On Mon, Mar 28, 2022 at 10:55 AM Andrei G. <andreigrav at gmail.com> wrote:

> Hi,
>
> How do I modify the field "Authorization: Digest username" in REGISTER
> packet?
>
> tried with uac_replace_to(,"user at asterisk1") and
> uac_replace_from(,"user at asterisk1") but it does affect only to and from
> headers not digest username
>
> thanks
> Andrei
>
>
> On Thu, Mar 24, 2022 at 4:20 AM Artiom Druz <artiom.druz at gmail.com> wrote:
>
>> Hello, Andrei.
>> You can do that by taking $tU (user from "To" header) and using some
>> transformation function. In your example can be used s.select function. (
>> https://www.opensips.org/Documentation/Script-Tran-3-2#toc7)
>>
>> Best regards,
>> Artiom Druz
>>
>> ср, 23 мар. 2022 г., 21:11 Andrei G. <andreigrav at gmail.com>:
>>
>>> Hey guys,
>>>
>>> I successfully tested opensips with mid_registrar for one domain
>>>
>>> Is it possible to use opensips in front of 2 asterisk boxes and redirect
>>> registrations based on a prefix username, not domain prefix?
>>>
>>> Something like
>>> asterisk1-user at mid-registrar.domain - where opensips manage
>>> registration for user at asterisk1
>>> asterisk2-user at mid-registrar.domain - where opensips manage
>>> registration for user at asterisk2
>>>
>>> Regards
>>> Andrei G.
>>>
>>> _______________________________________________
>>> 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 chester at zigbang.com  Tue Mar 29 03:40:05 2022
From: chester at zigbang.com (=?UTF-8?B?7J206riw7JuQ?=)
Date: Tue, 29 Mar 2022 12:40:05 +0900
Subject: [OpenSIPS-Users] strange INVITE transmission
In-Reply-To: <CAN0jgiNzT3xEKfNMkYLtAEbF9Z_EMxz7D+sWWLFXr_eBWoBx0Q@mail.gmail.com>
References: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>
 <CAN0jgiNzT3xEKfNMkYLtAEbF9Z_EMxz7D+sWWLFXr_eBWoBx0Q@mail.gmail.com>
Message-ID: <CA+-iJjZ_iqDcEJqEYuwdX=-2+Kc7cr1OM4hj_qUV0C-XpawU3g@mail.gmail.com>

Hi Daniel and opensips users goup,
Thank you for answering my questions.

I missed to inform you about the opensips version I'm trying. It is 3.2.5.

After modifying config, opensips does not respond for the REGISTER requests
from phones.
Actually I already tried the following
https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/
article but I have the same problem - no response for REGISTERs.

Is there any way to know why opensips ignores or does not respond for
REGISTERs?
Please find my new opensips.cfg that Diniel's advice is applied.


Thank you

Regards
Kiwon

2022년 3월 28일 (월) 오후 10:47, Daniel Zanutti <daniel.zanutti at gmail.com>님이 작성:

> Hi Kiwon
>
> You need to handle NAT scenarios. Try putting this code on line 254, right
> after "t_check_trans()":
>
> if (nat_uac_test("7"))
> {
>   #nathelper
>   if(is_method("REGISTER"))
>     fix_nated_register();
>   else
>     fix_nated_contact();
>   xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
> ID=$ci\n");
> }
>
> You also need to enable nathelper module. The reason is that you need to
> use the public IP/Port that sent data to opensips and ignore the Contact.
>
>
> On Mon, Mar 28, 2022 at 5:20 AM 이기원 <chester at zigbang.com> wrote:
>
>> Hi guys, I'm trying to introduce opensips into my company these days.
>>
>> I'm testing with many other phones. During the test, I had a very strange
>> symptom.
>>
>> Please, take look at the picture below first. (Please find my screenshot
>> from the link below)
>> A. The most left one 14.52.252.236 is a phone (with hardware).
>> B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
>> 58.79.209.75
>> C. 175.223.34.31 is a zoiper softphone which is working on iphone
>> D. 192.168.10.187 is the private IP address of A - a phone which is
>> the most left one (14.52.252.236)
>>
>>
>>
>>
>> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>>
>> As you can see, C is calling A but opensips transmits the INVITE rqeuset
>> to A's private IP address.
>> INVITE request should be transmitted A's public IP address - In this
>> scenario 14.52.252.236 (A)
>>
>> I also attach my opensips.cfg. What's wrong with me?
>>
>> Thank you
>>
>>
>> Regards
>> Kiwon
>> _______________________________________________
>> 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 daniel.zanutti at gmail.com  Tue Mar 29 04:04:57 2022
From: daniel.zanutti at gmail.com (Daniel Zanutti)
Date: Tue, 29 Mar 2022 01:04:57 -0300
Subject: [OpenSIPS-Users] strange INVITE transmission
In-Reply-To: <CA+-iJjZ_iqDcEJqEYuwdX=-2+Kc7cr1OM4hj_qUV0C-XpawU3g@mail.gmail.com>
References: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>
 <CAN0jgiNzT3xEKfNMkYLtAEbF9Z_EMxz7D+sWWLFXr_eBWoBx0Q@mail.gmail.com>
 <CA+-iJjZ_iqDcEJqEYuwdX=-2+Kc7cr1OM4hj_qUV0C-XpawU3g@mail.gmail.com>
Message-ID: <CAN0jgiOcji+2UFGuLi_3m+mxZhZhN7kSeq9d=DZVedpm6cMBig@mail.gmail.com>

Hi Kiwon

You applied at beginning, before loose_route. Not sure it gonna work this
way, i sent you exactly line where to put the code.
Move this:
 if (nat_uac_test(23)) {
if (is_method("REGISTER")) {
fix_nated_register();
setbflag("NAT");
} else {
fix_nated_contact();
setflag("NAT");
}
}

After:
t_check_trans();

Anyway, I suggest you add some log to confirm messages are coming. Put this
line right after main route:
xlog("L_ERR","MESSAGE RECEIVED $rm [$fu/$tu/$ru/$ci/$si]");

There's no big deal on Opensips, everything comes to main route, internal
transaction responses comes from specific routes. (
https://www.opensips.org/Documentation/Script-Routes-3-1)

I have some spare time tomorrow, send me a direct message if you need help.


On Tue, Mar 29, 2022 at 12:43 AM 이기원 <chester at zigbang.com> wrote:

> Hi Daniel and opensips users goup,
> Thank you for answering my questions.
>
> I missed to inform you about the opensips version I'm trying. It is 3.2.5.
>
> After modifying config, opensips does not respond for the REGISTER
> requests from phones.
> Actually I already tried the following
> https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/
> article but I have the same problem - no response for REGISTERs.
>
> Is there any way to know why opensips ignores or does not respond for
> REGISTERs?
> Please find my new opensips.cfg that Diniel's advice is applied.
>
>
> Thank you
>
> Regards
> Kiwon
>
> 2022년 3월 28일 (월) 오후 10:47, Daniel Zanutti <daniel.zanutti at gmail.com>님이 작성:
>
>> Hi Kiwon
>>
>> You need to handle NAT scenarios. Try putting this code on line 254,
>> right after "t_check_trans()":
>>
>> if (nat_uac_test("7"))
>> {
>>   #nathelper
>>   if(is_method("REGISTER"))
>>     fix_nated_register();
>>   else
>>     fix_nated_contact();
>>   xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
>> ID=$ci\n");
>> }
>>
>> You also need to enable nathelper module. The reason is that you need to
>> use the public IP/Port that sent data to opensips and ignore the Contact.
>>
>>
>> On Mon, Mar 28, 2022 at 5:20 AM 이기원 <chester at zigbang.com> wrote:
>>
>>> Hi guys, I'm trying to introduce opensips into my company these days.
>>>
>>> I'm testing with many other phones. During the test, I had a very
>>> strange symptom.
>>>
>>> Please, take look at the picture below first. (Please find my screenshot
>>> from the link below)
>>> A. The most left one 14.52.252.236 is a phone (with hardware).
>>> B. 10.0.0.177 is proxy which is woking on cloud and its external IP is
>>> 58.79.209.75
>>> C. 175.223.34.31 is a zoiper softphone which is working on iphone
>>> D. 192.168.10.187 is the private IP address of A - a phone which is
>>> the most left one (14.52.252.236)
>>>
>>>
>>>
>>>
>>> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>>>
>>> As you can see, C is calling A but opensips transmits the INVITE rqeuset
>>> to A's private IP address.
>>> INVITE request should be transmitted A's public IP address - In this
>>> scenario 14.52.252.236 (A)
>>>
>>> I also attach my opensips.cfg. What's wrong with me?
>>>
>>> Thank you
>>>
>>>
>>> Regards
>>> Kiwon
>>> _______________________________________________
>>> 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 kwem at gmx.de  Tue Mar 29 15:01:49 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Tue, 29 Mar 2022 17:01:49 +0200
Subject: [OpenSIPS-Users] Calling multiple contacts
Message-ID: <fd3a536eb41b101b9835ed27f0d569cab0a0b6a5.camel@gmx.de>

Hi *,

I am new to the list and I have a problem of understanding that
unfortunately I could not solve with Google and reading the
documentation.

My setup:
Phones register to OpenSIPs. Calls from the phones are transferred to a
PBX via t_relay. This part is working fine.

Requests from the PBX arrive in a route block to "lookup()". Now it is
so that some AOR are found with multiple contacts. t_relay() directs
these to the correct targets. Phones are ringing in parallel.

My problem: In some cases, manipulations have to be done on only some
of the outgoing requests. Think of one contact reachable by udp, others
by tcp or tls. I thought of using record_route_preset in the
branch_route. But this leads to an error (calling rr twice is not
possible).

Do You have any hints or examples for such a use case?

Thanks in advance,

Karsten



From bogdan at opensips.org  Wed Mar 30 07:52:06 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Wed, 30 Mar 2022 10:52:06 +0300
Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training 2022
Message-ID: <98b76556-f672-470c-5a90-8ac2512df014@opensips.org>

<https://opensips.org/training/OpenSIPS_eBootcamp_2021>


    23rd May - 03rd June 2022,


        online, worldwide



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*

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<https://opensips.org/training/OpenSIPS_eBootcamp_2022> and improve your 
OpenSIPS skills - an in-cloud training, a ten days, 4 hours per day (40 
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All the knowledge transferred to the students will be strongly backed up 
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OpenSIPS. The training is structured to be offer 50% / 50% between the 
theoretical and practical sessions.

Check Syllabus 
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of April 2022*/, so do not miss the opportunity. The number of seats is 
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discount is also available - grab your work mate and start learning more 
OpenSIPS together .
.

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<https://opensips.org/training/OpenSIPS_eBootcamp_2022/#mu-pricing>

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Companies that use OpenSIPS waste time and money when they don't have a 
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Any questions? do not hesitate to contact us <mailto:bootcamp at opensips.org>!

------------------------------------------------------------------------
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Project.
If you do not want to receive any more news, please email to unsubscribe 
<mailto:news-leave at lists.opensips.org?Subject=Unsubscribe>.


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

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From kwem at gmx.de  Wed Mar 30 18:04:09 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Wed, 30 Mar 2022 20:04:09 +0200
Subject: [OpenSIPS-Users] Calling multiple contacts
In-Reply-To: <fd3a536eb41b101b9835ed27f0d569cab0a0b6a5.camel@gmx.de>
References: <fd3a536eb41b101b9835ed27f0d569cab0a0b6a5.camel@gmx.de>
Message-ID: <43174c793d1023a4cd3eb53dd5b40fd9eb141b94.camel@gmx.de>

Hi,

Am Dienstag, dem 29.03.2022 um 17:01 +0200 schrieb Karsten Wemheuer:
> Hi *,
>
> I am new to the list and I have a problem of understanding that
> unfortunately I could not solve with Google and reading the
> documentation.
>
> My setup:
> Phones register to OpenSIPs. Calls from the phones are transferred to
> a
> PBX via t_relay. This part is working fine.
>
> Requests from the PBX arrive in a route block to "lookup()". Now it
> is
> so that some AOR are found with multiple contacts. t_relay() directs
> these to the correct targets. Phones are ringing in parallel.
>
> My problem: In some cases, manipulations have to be done on only some
> of the outgoing requests. Think of one contact reachable by udp,
> others
> by tcp or tls. I thought of using record_route_preset in the
> branch_route. But this leads to an error (calling rr twice is not
> possible).
>
> Do You have any hints or examples for such a use case?
>
> Thanks in advance,
>
> Karsten

to make the whole thing a bit more concrete, here is the section from
the configuration:

route[TOPHONES] {
  if ( !lookup("location") ) {
    sl_send_reply(404, "Not Found");
    exit;
  }

  t_on_branch("AST2PHONE");

  if ( $rm == "CANCEL" ) {
    if (!t_relay(8)) {
      sl_reply_error();
    }
    exit;
  }

  if (!t_relay()) {
    sl_reply_error();
  }
}

branch_route[AST2PHONE] {
  t_on_reply("PHONE_REPlY");
}

In some cases lookup() returns multiple responses. The entries found
must be partially reached via UDP or TLS or via NAT. Should I use
record_route? Where could this happen?

Thanks for any hints!

Best regards,

Karsten



From johan at democon.be  Thu Mar 31 08:23:35 2022
From: johan at democon.be (johan)
Date: Thu, 31 Mar 2022 10:23:35 +0200
Subject: [OpenSIPS-Users] simple way to see if a call is established.
Message-ID: <db5af378-d363-c81f-9f90-18c9c11cc79c@democon.be>

Hello,


is there a simple way to check in the script if a call has received it's
ACK ?  

or where exactly does ACK enter the script ?


wkr,



From kwem at gmx.de  Thu Mar 31 10:41:22 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Thu, 31 Mar 2022 12:41:22 +0200
Subject: [OpenSIPS-Users] simple way to see if a call is established.
In-Reply-To: <db5af378-d363-c81f-9f90-18c9c11cc79c@democon.be>
References: <db5af378-d363-c81f-9f90-18c9c11cc79c@democon.be>
Message-ID: <530c6473d4cf6f4c401b256755bac6a7615053ab.camel@gmx.de>

Hi Johan,

Am Donnerstag, dem 31.03.2022 um 10:23 +0200 schrieb johan:
> Hello,
>
> is there a simple way to check in the script if a call has received
> it's
> ACK ?
>
> or where exactly does ACK enter the script ?
>
I don't know how to check this, but as AFAIK the ACK enters the main
routing block as a request.

HTH,

Karsten



From kwem at gmx.de  Thu Mar 31 11:50:19 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Thu, 31 Mar 2022 13:50:19 +0200
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
Message-ID: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>

Hi*,

I have a understanding problem regarding branches and call forking.
A call from a PBX is to be routed to phone(s) via OpenSIPS. The phones
are registered to OpenSIPs.

INVITE --> lookup ----> 1. Destination
                   |
                   \--> 2. Destination

When the call is terminated by the caller, the BYE request shall take
the same path. Currently, the BYE is sent from the PBX directly to the
Contact URI (which is not reachable by the PBX).

Is it possible to use record_route in the branch_route so that
different record route headers are used? Or is there another way?

Thanks in advance,

Karsten



From Ben.Newlin at genesys.com  Thu Mar 31 13:31:15 2022
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Thu, 31 Mar 2022 13:31:15 +0000
Subject: [OpenSIPS-Users] simple way to see if a call is established.
In-Reply-To: <530c6473d4cf6f4c401b256755bac6a7615053ab.camel@gmx.de>
References: <db5af378-d363-c81f-9f90-18c9c11cc79c@democon.be>
 <530c6473d4cf6f4c401b256755bac6a7615053ab.camel@gmx.de>
Message-ID: <DS7PR10MB5214259B217C282DD24F2BA295E19@DS7PR10MB5214.namprd10.prod.outlook.com>

If you are using the dialog module the ACK status is reflected in the dialog status [1].

[1] - https://opensips.org/docs/modules/3.2.x/dialog.html#pv_DLG_status

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Karsten Wemheuer <kwem at gmx.de>
Date: Thursday, March 31, 2022 at 6:43 AM
To: users at lists.opensips.org <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] simple way to see if a call is established.
EXTERNAL EMAIL - Please use caution with links and attachments

Hi Johan,

Am Donnerstag, dem 31.03.2022 um 10:23 +0200 schrieb johan:
> Hello,
>
> is there a simple way to check in the script if a call has received
> it's
> ACK ?
>
> or where exactly does ACK enter the script ?
>
I don't know how to check this, but as AFAIK the ACK enters the main
routing block as a request.

HTH,

Karsten


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From bogdan at opensips.org  Thu Mar 31 13:43:22 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 16:43:22 +0300
Subject: [OpenSIPS-Users] OpenSIPS user location clustering in
 Kubernetes behind AWS ELB
In-Reply-To: <AS8P195MB1287FD33FE5C11EDACB10D55BD029@AS8P195MB1287.EURP195.PROD.OUTLOOK.COM>
References: <AS8P195MB1287FD33FE5C11EDACB10D55BD029@AS8P195MB1287.EURP195.PROD.OUTLOOK.COM>
Message-ID: <22cf7bf4-ef21-1a60-4d07-2a26feb97a49@opensips.org>

Hi Jon,

So, what is the top question here? how to form the cluster considering 
the fact K8s gives yous PODs dynamic IPs?

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/1/22 7:03 PM, Jonathan Hunter wrote:
>
> Hi All,
>
> I am using openSIPS to load balance requests from Websocket clients 
> that are using sip.js and are connecting to AWS via WSS and the ELB, 
> the requests are then sent to freeSWITCH for some media functions and 
> then passed back.
>
> With a single openSIPS pod this works great, but I am now looking to 
> implement more than one openSIPS instance, so user location then 
> becomes an issue as I am routing using namespace.
>
> I have used the clusterer module when not in a dynamic k8s 
> environment, however I wondered if anyone else had tried/tested this?
>
> For me the neighbour node Ids may be an issue due to the dynamic 
> nature of pods, and the node_ids themselves, as I guess I can 
> prepopulate them but wondered if this had been tested out?
>
> Just want to make the openSIPS pods HA and scalable.
>
> Hope this makes sense? I am looking at other work arounds but thought 
> would put it out there.
>
> Thanks
>
> Jon
>
> Sent from Mail <https://go.microsoft.com/fwlink/?LinkId=550986> for 
> Windows
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Thu Mar 31 13:46:06 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 16:46:06 +0300
Subject: [OpenSIPS-Users] Event on DE-REGISTRATION
In-Reply-To: <CAFjCFzkLjP76C00Eao-F3TokcsQMP49Zm10LwK5sMdp9LcBDoA@mail.gmail.com>
References: <CAFjCFzkLjP76C00Eao-F3TokcsQMP49Zm10LwK5sMdp9LcBDoA@mail.gmail.com>
Message-ID: <10e9f3bb-8805-2c05-9a49-7d641c522ada@opensips.org>

Hi Callum,

Have you tried a quick hack like:


route[send_request]
{
     t_new_request();
}

event_route[E_UL_CONTACT_DELETE]
{
     route(send_request);
}


Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/3/22 7:12 PM, Callum Guy wrote:
> Hi All,
>
> I have configured my registrar with max_contacts 1, allowing 
> subsequent registrations from that contact to overwrite.
>
> I am looking to intercept the de-registration and send a message to 
> the losing contact. Ideally I would use the existing SIP connection to 
> send it before termination, the client devices are softphones under my 
> control, but I suppose the TCP connection to the previous device may 
> have been severed by the time the event is raised (I can test this 
> when I get there but advice is always welcome).
>
> Event capture is straightforward using E_UL_CONTACT_DELETE however I 
> am wondering if there is any way to generate SIP towards the leaving 
> contact at this point? t_new_request() is not available in event_route 
> and I wanted to reach out to the community for ideas before building 
> an external system to do this (i.e. rest_post to a simple HTTP client).
>
> Is there a neat way to do this within the config script or am I 
> already on the best-fit path here?
>
> Many thanks,
>
> Callum
>
>
>
>
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From bogdan at opensips.org  Thu Mar 31 13:49:45 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 16:49:45 +0300
Subject: [OpenSIPS-Users] Need some help in adding custome header in
 Cancel Request .
In-Reply-To: <16d51ba6-aab2-53e8-ff87-9e3566a632bf@opensips.org>
References: <CALf-Q44HKcB1S6pv0MvTt-mTA3GaRmJUiue4_hYsoYP=PfPK5g@mail.gmail.com>
 <CADKNvCp2caOj0n4CZ+Wyt4QFbww7-0BWMJMpORZ+LYBF_K8QKw@mail.gmail.com>
 <CALf-Q47=v9Eu7xsG-f7=d9rZBJ92+b=AZ_L+kKqGTzCj-dU3UQ@mail.gmail.com>
 <16d51ba6-aab2-53e8-ff87-9e3566a632bf@opensips.org>
Message-ID: <4e9d2d8d-a8f6-2cfe-4e04-fe0de7b87319@opensips.org>

Also take a look at this post 
https://blog.opensips.org/2016/11/15/cancel-request-and-reason-header/

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/9/22 6:14 PM, Răzvan Crainea wrote:
> Hi, Sasmita!
>
> I actually don't think local_route is run for CANCEL messages.
> You may want to try to add a more complex reason using 
> t_add_cancel_reason[1].
>
> [1] https://opensips.org/docs/modules/3.2.x/tm.html#idp6205808
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 3/9/22 12:07, Sasmita Panda wrote:
>> My call flow is like  below .
>>
>> A -- > INVITE TO OPENSIPS -- > B
>> A -- > CANCEL TO OPENSIPS -- > B
>>
>> While A sends Cancel to Opensips (adds a custom header ) . When 
>> Opensips generates Cancel for B it won't add the custom header .
>>
>> This can be done by local_route ?
>>
>> */Thanks & Regards/*
>> /Sasmita Panda/
>> /Senior Network Testing and Software Engineer/
>> /3CLogic , ph:07827611765/
>>
>>
>> On Wed, Mar 9, 2022 at 3:27 PM Nick Altmann <nick at altmann.pro 
>> <mailto:nick at altmann.pro>> wrote:
>>
>>     Hi,
>>
>>     If cancel request generated by opensips, then you can control it
>>     from local_route.
>>
>>     --
>>     Nick
>>
>>     ср, 9 мар. 2022 г. в 10:54, Sasmita Panda <spanda at 3clogic.com
>>     <mailto:spanda at 3clogic.com>>:
>>
>>         Hi All,
>>
>>         Cancel is generated Hop by Hop . When the Opensips server
>>         receives a Cancel , Then it generates Cancel for the next 
>> party .
>>
>>         I am adding a custom header in the Cancel request , but when the
>>         next Hop Cancel is getting generated that custom header is not
>>         getting added . How will I pass the custom header in the Cancel
>>         request to the destination ?
>>
>>         */Thanks & Regards/*
>>         /Sasmita Panda/
>>         /Senior Network Testing and Software Engineer/
>>         /3CLogic , ph:07827611765/
>>         _______________________________________________
>>         Users mailing list
>>         Users at lists.opensips.org <mailto:Users at lists.opensips.org>
>>         http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
>>
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Thu Mar 31 13:52:13 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 16:52:13 +0300
Subject: [OpenSIPS-Users] segfault on nathelper.so
In-Reply-To: <CANtQs+UfT=M+WUMc_S0D6vDnWnRano1SHm7-5hcq19uYEONYeQ@mail.gmail.com>
References: <CANtQs+UfT=M+WUMc_S0D6vDnWnRano1SHm7-5hcq19uYEONYeQ@mail.gmail.com>
Message-ID: <807e36bb-d81d-842e-1d64-40a7fe343502@opensips.org>

Hi Alex,

Could you post a full backtrace?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/11/22 7:59 PM, Алексей Низиенко wrote:
> Good day!
> I have a trouble in opensips 3.2.5
>
> version: opensips 3.2.5 (x86_64/linux)
> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
> MAX_URI_SIZE 1024, BUF_SIZE 65535
> poll method support: poll, epoll, sigio_rt, select.
> main.c compiled on  with gcc 8
>
> uname -a
> Linux wsip 4.19.0-18-amd64 #1 SMP Debian 4.19.208-1 (2021-09-29)
> x86_64 GNU/Linux
>
> lsb_release -a
> No LSB modules are available.
> Distributor ID: Debian
> Description:    Debian GNU/Linux 10 (buster)
> Release:        10
> Codename:       buster
>
>
> [New LWP 7934]
> [Thread debugging using libthread_db enabled]
> Using host libthread_db library "/lib/x86_64-linux-gnu/libthread_db.so.1".
> Core was generated by `/usr/sbin/opensips -P
> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'.
> Program terminated with signal SIGSEGV, Segmentation fault.
> #0  0x00007f0937d67679 in remove_from_hash
> (cell=cell at entry=0x7f0937f11e70) at nh_table.c:171
> 171     nh_table.c: No such file or directory
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Thu Mar 31 13:55:17 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 16:55:17 +0300
Subject: [OpenSIPS-Users] suggested sql database structure
In-Reply-To: <48adf4d4-1c46-aa7d-ca19-fd531c38d417@voipplus.net>
References: <48adf4d4-1c46-aa7d-ca19-fd531c38d417@voipplus.net>
Message-ID: <f8b753d2-f684-eb68-f6b0-d0e5cc3689e7@opensips.org>

Hi Marcin,

Maybe you should consider different approaches for the tables you need 
as Read-Only mode versus the tables needing write access. Or just go for 
cluster DB solution, like percona or galera.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/6/22 6:20 PM, Marcin Groszek wrote:
>     I would like to deploy opensips in multiple geographical locations 
> and wondering if anyone has any suggestions on sql database structure.
>
> dialog would be local to each instant/pop
> dr_* tables are read only with possibility of dr_gateways probing
> permissions read only
> acc write only
> avpops read/write.
>
> Ideally I would like to see individual pops continue to operate even 
> if routing to other locations is interrupted.
> should I use a single centralized database , or perhaps a 
> master/master cluster over public internet; testing of it was not very 
> successful.
>
> Any suggestions would be appreciated.
>
>



From bogdan at opensips.org  Thu Mar 31 14:06:11 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:06:11 +0300
Subject: [OpenSIPS-Users] tlt_mgm module - any way to pass cert/key as
 parameter for outgoing connection?
In-Reply-To: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>
References: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>
Message-ID: <6621cfda-ef6b-b4b4-7d26-128423a533b5@opensips.org>

Hi Yury,

I'm afraid this is not possible (to fetch the cert from an external 
source at runtime). A dirty hack may be to (1) do the rest and fetch the 
cert + key,  (2) to insert into (from script) into the tls_mgm db table 
and (3) fire an MI tls_reload cmd (from script) via the mi() script 
function [1]

[1] https://opensips.org/html/docs/modules/3.2.x/mi_script.html#func_mi

and yeah, I know, it is ugly :(

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/15/22 1:45 PM, Yury Kirsanov wrote:
> Hi,
> I've got a question, is there any way to pass SSL certificate and key 
> as a parameter to the tls_mgm module during script execution? For 
> example, first I do a REST request to our REST API server which 
> returns me all required parameters including certificate and key. Then 
> I'd like to use this response as a client certificate for outgoing 
> connection to some TLS-enabled server. Is there any way to do that? I 
> know I can use DB module and select a client certificate using avp 
> variable, but that's not convenient as it requires tls_reload MI 
> command each time the DB is updated.
>
> Thanks and best regards,
> Yury.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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From bogdan at opensips.org  Thu Mar 31 14:27:52 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:27:52 +0300
Subject: [OpenSIPS-Users] OpenSIPS timers
In-Reply-To: <CAND0Lkvi=YXr_puy3ewEBWhUHfpMJADqZB+oyMPd=3SUVQeaDw@mail.gmail.com>
References: <CAND0Lkvi=YXr_puy3ewEBWhUHfpMJADqZB+oyMPd=3SUVQeaDw@mail.gmail.com>
Message-ID: <f809e267-77f0-dc22-ea1a-288ac056323b@opensips.org>

Hi Ovidiu,

As warnings from the timer_ticker, do you get only for the tm-utimer 
task ? I'm asking as the key question here is where the bottleneck is : 
in the whole "timer" subsystem, or in the tm-utimer task only?

The TM "timer_partitions" creates multiple parallel timer lists, to 
avoid having large "amounts" of transactions handled at a moment in a 
single tm-utimer task (but rather split/partition the whole of amount of 
handled transactions into smaller chunks, to be handled one at a time in 
the timer task.

The "timer_workers" creates  more than one dedicated processes for 
handling the timer tasks (so scales up the timer sub-system).

If you get warnings only on tm-utimer, I suspect the bottleneck is TM 
related, mainly on performing re-transmissions (that's what that task is 
doing). So the increasing the timer-partitions should be the way to help.

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/24/22 12:54 AM, Ovidiu Sas wrote:
> Hello all,
>
> I'm working on tuning an opensips server. I get this pesky:
> WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled
> I was trying to get rid of them by playing with the tm
> timer_partitions parameter and the timer_workers core param.
> By increasing any of them doesn't increase performance.
> By increasing both of them, it actually decreases performance.
> The server is not at limit, the load on the UDP workers is around
> 50-60 with some spikes.
> I have around 3500+ cps sipp traffic.
>
> My understanding is that by increasing the number of timer_partitions,
> we will have more procs walking in parallel over the timer structures.
> If we have on timer structure, we have one proc walking over it.
> How is this working for two timer structures? What is the difference
> between the first and the second timer structure? Should we expect
> less work for each proc?
>
> For now, to reduce the occurrence of the warning log, I increased the
> timer interval for tm-utimer from 100ms to 200ms. This should be ok as
> the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set.
>
> Thanks,
> Ovidiu
>



From bogdan at opensips.org  Thu Mar 31 14:36:12 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:36:12 +0300
Subject: [OpenSIPS-Users] mid_registrar multi domains
In-Reply-To: <CALUe069Kibr2R0ZyWkjJWDqgb7Z51vbDUh0DG12y466XFvLCHA@mail.gmail.com>
References: <CALUe068xgro7AojNh8eP6Ad8MkCPG=uzA+5LWR4Xmjb+wv3nrg@mail.gmail.com>
 <CAEBOSMTxJGWaCCE_6BpdeVjG72hiGzm=J+cndMUVaCFaOONK8Q@mail.gmail.com>
 <CALUe069Kibr2R0ZyWkjJWDqgb7Z51vbDUh0DG12y466XFvLCHA@mail.gmail.com>
Message-ID: <807cac6b-7b5e-728f-7dc5-fff07b00e685@opensips.org>

Hi Andrei,

You MUST NOT change the auth username as you will break the whole 
authentication process. The username is part of alg. for computing the 
auth result. So, if you change the auth uername, the auth result will 
become invalid.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/28/22 10:55 AM, Andrei G. wrote:
> Hi,
>
> How do I modify the field "Authorization: Digest username" in REGISTER 
> packet?
>
> tried with uac_replace_to(,"user at asterisk1") and 
> uac_replace_from(,"user at asterisk1") but it does affect only to and 
> from headers not digest username
>
> thanks
> Andrei
>
>
>
>
> On Thu, Mar 24, 2022 at 4:20 AM Artiom Druz <artiom.druz at gmail.com 
> <mailto:artiom.druz at gmail.com>> wrote:
>
>     Hello, Andrei.
>     You can do that by taking $tU (user from "To" header) and using
>     some transformation function. In your example can be used s.select
>     function.
>     (https://www.opensips.org/Documentation/Script-Tran-3-2#toc7
>     <https://www.opensips.org/Documentation/Script-Tran-3-2#toc7>)
>
>     Best regards,
>     Artiom Druz
>
>     ср, 23 мар. 2022 г., 21:11 Andrei G. <andreigrav at gmail.com
>     <mailto:andreigrav at gmail.com>>:
>
>         Hey guys,
>
>         I successfully tested opensips with mid_registrar for one domain
>
>         Is it possible to use opensips in front of 2 asterisk boxes
>         and redirect registrations based on a prefix username, not
>         domain prefix?
>
>         Something like
>         asterisk1-user at mid-registrar.domain - where opensips manage
>         registration for user at asterisk1
>         asterisk2-user at mid-registrar.domain - where opensips manage
>         registration for user at asterisk2
>
>         Regards
>         Andrei G.
>
>         _______________________________________________
>         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

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From bogdan at opensips.org  Thu Mar 31 14:42:47 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:42:47 +0300
Subject: [OpenSIPS-Users] Calling multiple contacts
In-Reply-To: <43174c793d1023a4cd3eb53dd5b40fd9eb141b94.camel@gmx.de>
References: <fd3a536eb41b101b9835ed27f0d569cab0a0b6a5.camel@gmx.de>
 <43174c793d1023a4cd3eb53dd5b40fd9eb141b94.camel@gmx.de>
Message-ID: <426de6b6-276a-a7e5-fa2e-2e062835039d@opensips.org>

Hi Karsten,

why using the record_route_preset()? it just make your life more 
complicated. Simply do before the t_relay() a record_route() and you 
should be done.

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/30/22 9:04 PM, Karsten Wemheuer wrote:
> Hi,
>
> Am Dienstag, dem 29.03.2022 um 17:01 +0200 schrieb Karsten Wemheuer:
>> Hi *,
>>
>> I am new to the list and I have a problem of understanding that
>> unfortunately I could not solve with Google and reading the
>> documentation.
>>
>> My setup:
>> Phones register to OpenSIPs. Calls from the phones are transferred to
>> a
>> PBX via t_relay. This part is working fine.
>>
>> Requests from the PBX arrive in a route block to "lookup()". Now it
>> is
>> so that some AOR are found with multiple contacts. t_relay() directs
>> these to the correct targets. Phones are ringing in parallel.
>>
>> My problem: In some cases, manipulations have to be done on only some
>> of the outgoing requests. Think of one contact reachable by udp,
>> others
>> by tcp or tls. I thought of using record_route_preset in the
>> branch_route. But this leads to an error (calling rr twice is not
>> possible).
>>
>> Do You have any hints or examples for such a use case?
>>
>> Thanks in advance,
>>
>> Karsten
> to make the whole thing a bit more concrete, here is the section from
> the configuration:
>
> route[TOPHONES] {
>    if ( !lookup("location") ) {
>      sl_send_reply(404, "Not Found");
>      exit;
>    }
>
>    t_on_branch("AST2PHONE");
>
>    if ( $rm == "CANCEL" ) {
>      if (!t_relay(8)) {
>        sl_reply_error();
>      }
>      exit;
>    }
>
>    if (!t_relay()) {
>      sl_reply_error();
>    }
> }
>
> branch_route[AST2PHONE] {
>    t_on_reply("PHONE_REPlY");
> }
>
> In some cases lookup() returns multiple responses. The entries found
> must be partially reached via UDP or TLS or via NAT. Should I use
> record_route? Where could this happen?
>
> Thanks for any hints!
>
> Best regards,
>
> Karsten
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Thu Mar 31 14:44:20 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:44:20 +0300
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
In-Reply-To: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
References: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
Message-ID: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>

Hi Karsten,

See my prev email, just to record_route() before the t_relay() for the 
initial INVITE. And the loose_route() stuff for whatever 
sequential/in-dialog requests.

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> Hi*,
>
> I have a understanding problem regarding branches and call forking.
> A call from a PBX is to be routed to phone(s) via OpenSIPS. The phones
> are registered to OpenSIPs.
>
> INVITE --> lookup ----> 1. Destination
>                     |
>                     \--> 2. Destination
>
> When the call is terminated by the caller, the BYE request shall take
> the same path. Currently, the BYE is sent from the PBX directly to the
> Contact URI (which is not reachable by the PBX).
>
> Is it possible to use record_route in the branch_route so that
> different record route headers are used? Or is there another way?
>
> Thanks in advance,
>
> Karsten
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Thu Mar 31 14:47:44 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 17:47:44 +0300
Subject: [OpenSIPS-Users] Attended call transfer in opensips with use of
 RTPengine
In-Reply-To: <f0fb2d68-6a0e-d3a2-b940-55d84e01c92b@softnet.si>
References: <b0215761-dc49-1b60-558d-418bd216b63e@softnet.si>
 <55ddf70b-10a8-6c1f-6d04-f81f3bce311a@softnet.si>
 <21daf27a-bfec-04d5-184c-55f1deeef1b2@opensips.org>
 <f0fb2d68-6a0e-d3a2-b940-55d84e01c92b@softnet.si>
Message-ID: <6abc8b48-06c8-2bff-edca-0398a087d423@opensips.org>

Hi Simon,

If you have OpenSIPS and a simple SIP proxy (not a B2B), you must not 
reply to the REFER, but to let it be routed as an in-dialog request to 
the other end-point, as that end-point will actually do the transfer 
(starting the new call).

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/2/22 12:02 PM, Simon Gajski wrote:
>
> Hi
>
> no, we don't use B2B on OpenSIPS  side. Is this the correctway to do it?
> The thing is that I don't know what would be the best way to implement 
> this with use of RTPengine.
> I found very little info available online.
>
> Call forwards that I manually set in DB (cfu, cfnr, cfb.....like we 
> were doing on last bootcamp) are working fine.
> Only issue is with answered call and then attempting to transfer it.
>
> BR
> Simon
>
>
>
> Bogdan-Andrei Iancu je 01.03.2022 ob 15:30 napisal:
>> Hi Simon,
>>
>> Do you use B2B on the OpenSIPS side ? Which entity is actually 
>> performing the transfer ?
>>
>> Regards,
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>    https://www.opensips-solutions.com
>> OpenSIPS eBootcamp
>>    https://www.opensips.org/Training/Bootcamp
>> On 2/24/22 1:54 PM, Simon Gajski via Users wrote:
>>>
>>> Hi
>>>
>>>
>>> I am using opensips 3.2 with rtpengine on same server and trying to 
>>> achieve attended call transfer.
>>>
>>> In theory, I'm trying to do:
>>> 1. A calls B...and B answers
>>> 2. B puts A on hold (MOH is played from RTPengine)
>>> 3. B calls C...and C answers
>>>
>>> Now the funny part:
>>> B tries to transfer A to C and sends REFER to opensips
>>> In opensips I responds with 202 Accepted and B gets disconnected.
>>>
>>> However A and C don't get connected together
>>> A still receives MOH and C has no voice
>>>
>>> We have another installation of opensips where REFER handles 
>>> Freeswitch, and there such type of transfer is working fine.
>>>
>>> Can someone help me how to handle such call behaviour in opensips 
>>> with RTPengine?
>>>
>>>
>>> relevant part of code:
>>>
>>> route[handle_sequential]{
>>> ...
>>> if(is_method("REFER")) {
>>>         xlog("[IN_DIALOG] [$rm] Transfer from $fu to $tu");
>>>         send_reply(202, "Accepted");
>>>
>>>         #what next?
>>>
>>>         exit;
>>>     }
>>> ...
>>> }
>>>
>>>
>>> Thank you!
>>>
>>> Simon
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>

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From john.quick at smartvox.co.uk  Thu Mar 31 14:49:22 2022
From: john.quick at smartvox.co.uk (John Quick)
Date: Thu, 31 Mar 2022 15:49:22 +0100
Subject: [OpenSIPS-Users] OpenSIPS user location clustering in
	Kubernetes	behind AWS ELB
Message-ID: <001101d8450e$861dfac0$9259f040$@smartvox.co.uk>

Jon,

I worked on a project like this a few years ago. Unfortunately, it was
terminated before getting into production.
However, I presented a paper at the OpenSIPS Summit and if you have 30
minutes to spare you could watch the recording and see if there is anything
useful there.
Follow this link to get there.
https://www.smartvox.co.uk/2019/06/using-opensips-in-docker/

In theory, I'd be happy to answer any emailed questions if I can, but it's
all rather hazy now.

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




From ialex.gu at gmail.com  Thu Mar 31 15:02:53 2022
From: ialex.gu at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdC10Lkg0J3QuNC30LjQtdC90LrQvg==?=)
Date: Thu, 31 Mar 2022 18:02:53 +0300
Subject: [OpenSIPS-Users] segfault on nathelper.so
In-Reply-To: <807e36bb-d81d-842e-1d64-40a7fe343502@opensips.org>
References: <CANtQs+UfT=M+WUMc_S0D6vDnWnRano1SHm7-5hcq19uYEONYeQ@mail.gmail.com>
 <807e36bb-d81d-842e-1d64-40a7fe343502@opensips.org>
Message-ID: <CANtQs+Uz9K+Ys5Rqq=c+vc0_y3gv3LLqBjv+HrTLetAx6fqhVA@mail.gmail.com>

I believe the error went away after making changes to the routing.
rtpengine_delete() fixed it. I think so. Thanks for your reply,
Bogdan!

2022-03-31 16:52 GMT+03:00, Bogdan-Andrei Iancu <bogdan at opensips.org>:
> Hi Alex,
>
> Could you post a full backtrace?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/11/22 7:59 PM, Алексей Низиенко wrote:
>> Good day!
>> I have a trouble in opensips 3.2.5
>>
>> version: opensips 3.2.5 (x86_64/linux)
>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>> poll method support: poll, epoll, sigio_rt, select.
>> main.c compiled on  with gcc 8
>>
>> uname -a
>> Linux wsip 4.19.0-18-amd64 #1 SMP Debian 4.19.208-1 (2021-09-29)
>> x86_64 GNU/Linux
>>
>> lsb_release -a
>> No LSB modules are available.
>> Distributor ID: Debian
>> Description:    Debian GNU/Linux 10 (buster)
>> Release:        10
>> Codename:       buster
>>
>>
>> [New LWP 7934]
>> [Thread debugging using libthread_db enabled]
>> Using host libthread_db library
>> "/lib/x86_64-linux-gnu/libthread_db.so.1".
>> Core was generated by `/usr/sbin/opensips -P
>> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'.
>> Program terminated with signal SIGSEGV, Segmentation fault.
>> #0  0x00007f0937d67679 in remove_from_hash
>> (cell=cell at entry=0x7f0937f11e70) at nh_table.c:171
>> 171     nh_table.c: No such file or directory
>>
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>


From bogdan at opensips.org  Thu Mar 31 15:11:20 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 18:11:20 +0300
Subject: [OpenSIPS-Users] segfault on nathelper.so
In-Reply-To: <CANtQs+Uz9K+Ys5Rqq=c+vc0_y3gv3LLqBjv+HrTLetAx6fqhVA@mail.gmail.com>
References: <CANtQs+UfT=M+WUMc_S0D6vDnWnRano1SHm7-5hcq19uYEONYeQ@mail.gmail.com>
 <807e36bb-d81d-842e-1d64-40a7fe343502@opensips.org>
 <CANtQs+Uz9K+Ys5Rqq=c+vc0_y3gv3LLqBjv+HrTLetAx6fqhVA@mail.gmail.com>
Message-ID: <5c872470-51ed-37fd-7088-4b984bd7add3@opensips.org>

I see. But what you did is to avoid the bug, not to fix it. So, if you 
still have the full backtrace, it will be helpful for us to actually fix 
it ;)

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/31/22 6:02 PM, Алексей Низиенко wrote:
> I believe the error went away after making changes to the routing.
> rtpengine_delete() fixed it. I think so. Thanks for your reply,
> Bogdan!
>
> 2022-03-31 16:52 GMT+03:00, Bogdan-Andrei Iancu <bogdan at opensips.org>:
>> Hi Alex,
>>
>> Could you post a full backtrace?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>     https://www.opensips-solutions.com
>> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>>     https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 3/11/22 7:59 PM, Алексей Низиенко wrote:
>>> Good day!
>>> I have a trouble in opensips 3.2.5
>>>
>>> version: opensips 3.2.5 (x86_64/linux)
>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
>>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
>>> MAX_URI_SIZE 1024, BUF_SIZE 65535
>>> poll method support: poll, epoll, sigio_rt, select.
>>> main.c compiled on  with gcc 8
>>>
>>> uname -a
>>> Linux wsip 4.19.0-18-amd64 #1 SMP Debian 4.19.208-1 (2021-09-29)
>>> x86_64 GNU/Linux
>>>
>>> lsb_release -a
>>> No LSB modules are available.
>>> Distributor ID: Debian
>>> Description:    Debian GNU/Linux 10 (buster)
>>> Release:        10
>>> Codename:       buster
>>>
>>>
>>> [New LWP 7934]
>>> [Thread debugging using libthread_db enabled]
>>> Using host libthread_db library
>>> "/lib/x86_64-linux-gnu/libthread_db.so.1".
>>> Core was generated by `/usr/sbin/opensips -P
>>> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg'.
>>> Program terminated with signal SIGSEGV, Segmentation fault.
>>> #0  0x00007f0937d67679 in remove_from_hash
>>> (cell=cell at entry=0x7f0937f11e70) at nh_table.c:171
>>> 171     nh_table.c: No such file or directory
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>



From kwem at gmx.de  Thu Mar 31 15:22:53 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Thu, 31 Mar 2022 17:22:53 +0200
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
In-Reply-To: <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>
References: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
 <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>
Message-ID: <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>

Hi Bogdan-Andrei,

many thanks for Your help.

I tried with record_route. It doesn't work for me, as I set
"advertised_address" and "advertised_port" to the natted address of the
(only) interface. I wasn't able to avoid this. It seemed to be required
to be able to reflect the path "phone -> proxy -> pbx".

I removed the "advertised"-stuff and checked again the call with
record_route. Now this seems to work.

I think, I have to fix the other call flow to avoid the global setting
of the advertised address and port.

Best regards,

Karsten

Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
Iancu:
> Hi Karsten,
>
> See my prev email, just to record_route() before the t_relay() for
> the
> initial INVITE. And the loose_route() stuff for whatever
> sequential/in-dialog requests.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> > Hi*,
> >
> > I have a understanding problem regarding branches and call forking.
> > A call from a PBX is to be routed to phone(s) via OpenSIPS. The
> > phones
> > are registered to OpenSIPs.
> >
> > INVITE --> lookup ----> 1. Destination
> >                     |
> >                     \--> 2. Destination
> >
> > When the call is terminated by the caller, the BYE request shall
> > take
> > the same path. Currently, the BYE is sent from the PBX directly to
> > the
> > Contact URI (which is not reachable by the PBX).
> >
> > Is it possible to use record_route in the branch_route so that
> > different record route headers are used? Or is there another way?
> >
> > Thanks in advance,
> >
> > Karsten
> >
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From y.kirsanov at gmail.com  Thu Mar 31 15:23:44 2022
From: y.kirsanov at gmail.com (Yury Kirsanov)
Date: Fri, 1 Apr 2022 02:23:44 +1100
Subject: [OpenSIPS-Users] tlt_mgm module - any way to pass cert/key as
 parameter for outgoing connection?
In-Reply-To: <6621cfda-ef6b-b4b4-7d26-128423a533b5@opensips.org>
References: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>
 <6621cfda-ef6b-b4b4-7d26-128423a533b5@opensips.org>
Message-ID: <CAD1_seums3Gj6AhtA7PbLsrxoe=LHSmqB-XdLF4UDwraDhMLHg@mail.gmail.com>

Hi Bogdan,
Thanks, that's a good idea! Hope one day we will have the ability to select
certificates from AVPs in script!

Best regards,
Yury.

On Fri, Apr 1, 2022 at 1:06 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Yury,
>
> I'm afraid this is not possible (to fetch the cert from an external source
> at runtime). A dirty hack may be to (1) do the rest and fetch the cert +
> key,  (2) to insert into (from script) into the tls_mgm db table and (3)
> fire an MI tls_reload cmd (from script) via the mi() script function [1]
>
> [1] https://opensips.org/html/docs/modules/3.2.x/mi_script.html#func_mi
>
> and yeah, I know, it is ugly :(
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>   https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/15/22 1:45 PM, Yury Kirsanov wrote:
>
> Hi,
> I've got a question, is there any way to pass SSL certificate and key as a
> parameter to the tls_mgm module during script execution? For example, first
> I do a REST request to our REST API server which returns me all required
> parameters including certificate and key. Then I'd like to use this
> response as a client certificate for outgoing connection to some
> TLS-enabled server. Is there any way to do that? I know I can use DB module
> and select a client certificate using avp variable, but that's not
> convenient as it requires tls_reload MI command each time the DB is updated.
>
> Thanks and best regards,
> Yury.
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From bogdan at opensips.org  Thu Mar 31 15:25:15 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 18:25:15 +0300
Subject: [OpenSIPS-Users] tlt_mgm module - any way to pass cert/key as
 parameter for outgoing connection?
In-Reply-To: <CAD1_seums3Gj6AhtA7PbLsrxoe=LHSmqB-XdLF4UDwraDhMLHg@mail.gmail.com>
References: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>
 <6621cfda-ef6b-b4b4-7d26-128423a533b5@opensips.org>
 <CAD1_seums3Gj6AhtA7PbLsrxoe=LHSmqB-XdLF4UDwraDhMLHg@mail.gmail.com>
Message-ID: <e6cc1175-e7cd-6ed4-a42e-a59e526bbe9d@opensips.org>

Hi Yury,

You can open a feature request on github, so we can take this into 
consideration for the future releases ;)

Best regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/31/22 6:23 PM, Yury Kirsanov wrote:
> Hi Bogdan,
> Thanks, that's a good idea! Hope one day we will have the ability to 
> select certificates from AVPs in script!
>
> Best regards,
> Yury.
>
> On Fri, Apr 1, 2022 at 1:06 AM Bogdan-Andrei Iancu 
> <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>     Hi Yury,
>
>     I'm afraid this is not possible (to fetch the cert from an
>     external source at runtime). A dirty hack may be to (1) do the
>     rest and fetch the cert + key,  (2) to insert into (from script)
>     into the tls_mgm db table and (3) fire an MI tls_reload cmd (from
>     script) via the mi() script function [1]
>
>     [1]
>     https://opensips.org/html/docs/modules/3.2.x/mi_script.html#func_mi
>     <https://opensips.org/html/docs/modules/3.2.x/mi_script.html#func_mi>
>
>     and yeah, I know, it is ugly :(
>
>     Best regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
>     OpenSIPS eBootcamp 23rd May - 3rd June 2022
>        https://opensips.org/training/OpenSIPS_eBootcamp_2022/  <https://opensips.org/training/OpenSIPS_eBootcamp_2022/>
>
>     On 3/15/22 1:45 PM, Yury Kirsanov wrote:
>>     Hi,
>>     I've got a question, is there any way to pass SSL certificate and
>>     key as a parameter to the tls_mgm module during script execution?
>>     For example, first I do a REST request to our REST API server
>>     which returns me all required parameters including certificate
>>     and key. Then I'd like to use this response as a client
>>     certificate for outgoing connection to some TLS-enabled server.
>>     Is there any way to do that? I know I can use DB module and
>>     select a client certificate using avp variable, but that's not
>>     convenient as it requires tls_reload MI command each time the DB
>>     is updated.
>>
>>     Thanks and best regards,
>>     Yury.
>>
>>     _______________________________________________
>>     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 y.kirsanov at gmail.com  Thu Mar 31 15:26:51 2022
From: y.kirsanov at gmail.com (Yury Kirsanov)
Date: Fri, 1 Apr 2022 02:26:51 +1100
Subject: [OpenSIPS-Users] tlt_mgm module - any way to pass cert/key as
 parameter for outgoing connection?
In-Reply-To: <e6cc1175-e7cd-6ed4-a42e-a59e526bbe9d@opensips.org>
References: <CAD1_sesJSFOCmtWwsenr39s=gGMAsNbukcMZUBoJOugVHoWK7g@mail.gmail.com>
 <6621cfda-ef6b-b4b4-7d26-128423a533b5@opensips.org>
 <CAD1_seums3Gj6AhtA7PbLsrxoe=LHSmqB-XdLF4UDwraDhMLHg@mail.gmail.com>
 <e6cc1175-e7cd-6ed4-a42e-a59e526bbe9d@opensips.org>
Message-ID: <CAD1_setQr0fNvT6dzFKeAAbZ8uNPWW6dvTExjGs+2o+Vn=abyw@mail.gmail.com>

Hi Bogdan,
Thanks, will do!

Best regards,
Yury.

On Fri, Apr 1, 2022 at 2:25 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi Yury,
>
> You can open a feature request on github, so we can take this into
> consideration for the future releases ;)
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>   https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/31/22 6:23 PM, Yury Kirsanov wrote:
>
> Hi Bogdan,
> Thanks, that's a good idea! Hope one day we will have the ability to
> select certificates from AVPs in script!
>
> Best regards,
> Yury.
>
> On Fri, Apr 1, 2022 at 1:06 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi Yury,
>>
>> I'm afraid this is not possible (to fetch the cert from an external
>> source at runtime). A dirty hack may be to (1) do the rest and fetch the
>> cert + key,  (2) to insert into (from script) into the tls_mgm db table and
>> (3) fire an MI tls_reload cmd (from script) via the mi() script function [1]
>>
>> [1] https://opensips.org/html/docs/modules/3.2.x/mi_script.html#func_mi
>>
>> and yeah, I know, it is ugly :(
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>>   https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 3/15/22 1:45 PM, Yury Kirsanov wrote:
>>
>> Hi,
>> I've got a question, is there any way to pass SSL certificate and key as
>> a parameter to the tls_mgm module during script execution? For example,
>> first I do a REST request to our REST API server which returns me all
>> required parameters including certificate and key. Then I'd like to use
>> this response as a client certificate for outgoing connection to some
>> TLS-enabled server. Is there any way to do that? I know I can use DB module
>> and select a client certificate using avp variable, but that's not
>> convenient as it requires tls_reload MI command each time the DB is updated.
>>
>> Thanks and best regards,
>> Yury.
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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From bogdan at opensips.org  Thu Mar 31 15:53:33 2022
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 31 Mar 2022 18:53:33 +0300
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
In-Reply-To: <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>
References: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
 <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>
 <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>
Message-ID: <a4a33ec6-a84d-cce2-c47f-9deb58876451@opensips.org>

Hi Karsten,

You say the record_route() does not take into consideration the global 
advertising ??

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
OpenSIPS eBootcamp 23rd May - 3rd June 2022
   https://opensips.org/training/OpenSIPS_eBootcamp_2022/

On 3/31/22 6:22 PM, Karsten Wemheuer wrote:
> Hi Bogdan-Andrei,
>
> many thanks for Your help.
>
> I tried with record_route. It doesn't work for me, as I set
> "advertised_address" and "advertised_port" to the natted address of the
> (only) interface. I wasn't able to avoid this. It seemed to be required
> to be able to reflect the path "phone -> proxy -> pbx".
>
> I removed the "advertised"-stuff and checked again the call with
> record_route. Now this seems to work.
>
> I think, I have to fix the other call flow to avoid the global setting
> of the advertised address and port.
>
> Best regards,
>
> Karsten
>
> Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
> Iancu:
>> Hi Karsten,
>>
>> See my prev email, just to record_route() before the t_relay() for
>> the
>> initial INVITE. And the loose_route() stuff for whatever
>> sequential/in-dialog requests.
>>
>> Best regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>     https://www.opensips-solutions.com
>> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>>     https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>>
>> On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
>>> Hi*,
>>>
>>> I have a understanding problem regarding branches and call forking.
>>> A call from a PBX is to be routed to phone(s) via OpenSIPS. The
>>> phones
>>> are registered to OpenSIPs.
>>>
>>> INVITE --> lookup ----> 1. Destination
>>>                      |
>>>                      \--> 2. Destination
>>>
>>> When the call is terminated by the caller, the BYE request shall
>>> take
>>> the same path. Currently, the BYE is sent from the PBX directly to
>>> the
>>> Contact URI (which is not reachable by the PBX).
>>>
>>> Is it possible to use record_route in the branch_route so that
>>> different record route headers are used? Or is there another way?
>>>
>>> Thanks in advance,
>>>
>>> Karsten
>>>
>>>
>>> _______________________________________________
>>> Users mailing list
>>> Users at lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From kwem at gmx.de  Thu Mar 31 15:54:53 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Thu, 31 Mar 2022 17:54:53 +0200
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
In-Reply-To: <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>
References: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
 <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>
 <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>
Message-ID: <587e5e6335dece1cf51d7d9a117e0a86eba3db03.camel@gmx.de>

Hi,

using record_route(), lookup("location) and t_relay() seems to work,
but not in all cases.
I have the 3 use cases:
1) phones reachable in local network via UDP
2) phones reachable in local network via TLS
3) phones in public internet behind NAT via TLS

The last case works except if you end the call from the phone. The
Record-Route header contains the IP address of the LAN not the address
reachable by the phone. The BYE doesn't reach the proxy.

Is it possible to change this, maybe somewhere in the branch route?
(Sometimes a call is directed towards phones of cases (2) and (3) in
parallel).

Thanks in advance,

Karsten


Am Donnerstag, dem 31.03.2022 um 17:22 +0200 schrieb Karsten Wemheuer:
> Hi Bogdan-Andrei,
>
> many thanks for Your help.
>
> I tried with record_route. It doesn't work for me, as I set
> "advertised_address" and "advertised_port" to the natted address of
> the
> (only) interface. I wasn't able to avoid this. It seemed to be
> required
> to be able to reflect the path "phone -> proxy -> pbx".
>
> I removed the "advertised"-stuff and checked again the call with
> record_route. Now this seems to work.
>
> I think, I have to fix the other call flow to avoid the global
> setting
> of the advertised address and port.
>
> Best regards,
>
> Karsten
>
> Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
> Iancu:
> > Hi Karsten,
> >
> > See my prev email, just to record_route() before the t_relay() for
> > the
> > initial INVITE. And the loose_route() stuff for whatever
> > sequential/in-dialog requests.
> >
> > Best regards,
> >
> > Bogdan-Andrei Iancu
> >
> > OpenSIPS Founder and Developer
> >    https://www.opensips-solutions.com
> > OpenSIPS eBootcamp 23rd May - 3rd June 2022
> >    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
> >
> > On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> > > Hi*,
> > >
> > > I have a understanding problem regarding branches and call
> > > forking.
> > > A call from a PBX is to be routed to phone(s) via OpenSIPS. The
> > > phones
> > > are registered to OpenSIPs.
> > >
> > > INVITE --> lookup ----> 1. Destination
> > >                     |
> > >                     \--> 2. Destination
> > >
> > > When the call is terminated by the caller, the BYE request shall
> > > take
> > > the same path. Currently, the BYE is sent from the PBX directly
> > > to
> > > the
> > > Contact URI (which is not reachable by the PBX).
> > >
> > > Is it possible to use record_route in the branch_route so that
> > > different record route headers are used? Or is there another way?
> > >
> > > Thanks in advance,
> > >
> > > Karsten
> > >
> > >
> > > _______________________________________________
> > > Users mailing list
> > > Users at lists.opensips.org
> > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From kwem at gmx.de  Thu Mar 31 16:27:47 2022
From: kwem at gmx.de (Karsten Wemheuer)
Date: Thu, 31 Mar 2022 18:27:47 +0200
Subject: [OpenSIPS-Users] Call forking, branches, Record-routing
In-Reply-To: <a4a33ec6-a84d-cce2-c47f-9deb58876451@opensips.org>
References: <ecb36100594d59d3b503f47674b28907564d618c.camel@gmx.de>
 <45a50a6c-0aec-6862-3710-8dbf2085ec8e@opensips.org>
 <0239fc41a5d0265deeb86562ccc7e57d0a819050.camel@gmx.de>
 <a4a33ec6-a84d-cce2-c47f-9deb58876451@opensips.org>
Message-ID: <80730f412ba7559ccd1de5459fdf7ce154992cf4.camel@gmx.de>

Hi Bogdan-Andrei,

in case of global advertising is active and set to the natted address
the advertised address is used, but this leads to problems using phones
in the LAN.

As written in my other post: Without setting the advertise address and
port, I have a problem with the phones behind NAT. Is it possible to
manipulate the route before in a branch or something like that?

Regards,

Karsten

Am Donnerstag, dem 31.03.2022 um 18:53 +0300 schrieb Bogdan-Andrei
Iancu:
> Hi Karsten,
>
> You say the record_route() does not take into consideration the
> global
> advertising ??
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/31/22 6:22 PM, Karsten Wemheuer wrote:
> > Hi Bogdan-Andrei,
> >
> > many thanks for Your help.
> >
> > I tried with record_route. It doesn't work for me, as I set
> > "advertised_address" and "advertised_port" to the natted address of
> > the
> > (only) interface. I wasn't able to avoid this. It seemed to be
> > required
> > to be able to reflect the path "phone -> proxy -> pbx".
> >
> > I removed the "advertised"-stuff and checked again the call with
> > record_route. Now this seems to work.
> >
> > I think, I have to fix the other call flow to avoid the global
> > setting
> > of the advertised address and port.
> >
> > Best regards,
> >
> > Karsten
> >
> > Am Donnerstag, dem 31.03.2022 um 17:44 +0300 schrieb Bogdan-Andrei
> > Iancu:
> > > Hi Karsten,
> > >
> > > See my prev email, just to record_route() before the t_relay()
> > > for
> > > the
> > > initial INVITE. And the loose_route() stuff for whatever
> > > sequential/in-dialog requests.
> > >
> > > Best regards,
> > >
> > > Bogdan-Andrei Iancu
> > >
> > > OpenSIPS Founder and Developer
> > >     https://www.opensips-solutions.com
> > > OpenSIPS eBootcamp 23rd May - 3rd June 2022
> > >     https://opensips.org/training/OpenSIPS_eBootcamp_2022/
> > >
> > > On 3/31/22 2:50 PM, Karsten Wemheuer wrote:
> > > > Hi*,
> > > >
> > > > I have a understanding problem regarding branches and call
> > > > forking.
> > > > A call from a PBX is to be routed to phone(s) via OpenSIPS. The
> > > > phones
> > > > are registered to OpenSIPs.
> > > >
> > > > INVITE --> lookup ----> 1. Destination
> > > >                      |
> > > >                      \--> 2. Destination
> > > >
> > > > When the call is terminated by the caller, the BYE request
> > > > shall
> > > > take
> > > > the same path. Currently, the BYE is sent from the PBX directly
> > > > to
> > > > the
> > > > Contact URI (which is not reachable by the PBX).
> > > >
> > > > Is it possible to use record_route in the branch_route so that
> > > > different record route headers are used? Or is there another
> > > > way?
> > > >
> > > > Thanks in advance,
> > > >
> > > > Karsten
> > > >
> > > >
> > > > _______________________________________________
> > > > Users mailing list
> > > > Users at lists.opensips.org
> > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From hunterj91 at hotmail.com  Thu Mar 31 18:36:35 2022
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Thu, 31 Mar 2022 18:36:35 +0000
Subject: [OpenSIPS-Users] OpenSIPS user location clustering
	in	Kubernetes	behind AWS ELB
In-Reply-To: <001101d8450e$861dfac0$9259f040$@smartvox.co.uk>
References: <001101d8450e$861dfac0$9259f040$@smartvox.co.uk>
Message-ID: <AS8P195MB128799A9E58B043E89155653BDE19@AS8P195MB1287.EURP195.PROD.OUTLOOK.COM>

HI John,

Thank you for the response, and I very much enjoyed your presentation.

I almost have the solution working so will indeed keep you and community in the loop when this is the case.

Much appreciated  John thank you!  - Note your website is also a useful resource :)

Thanks again.

Jon

Sent from Mail<https://go.microsoft.com/fwlink/?LinkId=550986> for Windows

From: John Quick<mailto:john.quick at smartvox.co.uk>
Sent: 31 March 2022 15:51
To: users at lists.opensips.org<mailto:users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] OpenSIPS user location clustering in Kubernetes behind AWS ELB

Jon,

I worked on a project like this a few years ago. Unfortunately, it was
terminated before getting into production.
However, I presented a paper at the OpenSIPS Summit and if you have 30
minutes to spare you could watch the recording and see if there is anything
useful there.
Follow this link to get there.
https://nam12.safelinks.protection.outlook.com/?url=https%3A%2F%2Fwww.smartvox.co.uk%2F2019%2F06%2Fusing-opensips-in-docker%2F&amp;data=04%7C01%7C%7C80308007847f435c84e008da1325f25b%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637843350965546611%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&amp;sdata=RQhjV4Fp%2FI9iryXjn0ddp6lio9%2FtyPSTd3zybya36kY%3D&amp;reserved=0

In theory, I'd be happy to answer any emailed questions if I can, but it's
all rather hazy now.

John Quick
Smartvox Limited
Web: https://nam12.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.smartvox.co.uk%2F&amp;data=04%7C01%7C%7C80308007847f435c84e008da1325f25b%7C84df9e7fe9f640afb435aaaaaaaaaaaa%7C1%7C0%7C637843350965546611%7CUnknown%7CTWFpbGZsb3d8eyJWIjoiMC4wLjAwMDAiLCJQIjoiV2luMzIiLCJBTiI6Ik1haWwiLCJXVCI6Mn0%3D%7C3000&amp;sdata=h8LyfMxkZ3fsHq1N4BDVfJlyGcj4qb4825xpKz%2FU4Ig%3D&amp;reserved=0



_______________________________________________
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From osas at voipembedded.com  Thu Mar 31 21:31:33 2022
From: osas at voipembedded.com (Ovidiu Sas)
Date: Thu, 31 Mar 2022 17:31:33 -0400
Subject: [OpenSIPS-Users] OpenSIPS timers
In-Reply-To: <f809e267-77f0-dc22-ea1a-288ac056323b@opensips.org>
References: <CAND0Lkvi=YXr_puy3ewEBWhUHfpMJADqZB+oyMPd=3SUVQeaDw@mail.gmail.com>
 <f809e267-77f0-dc22-ea1a-288ac056323b@opensips.org>
Message-ID: <CAND0LktDFezOtyyfKcAikN30S3gRTp_2HAjXqfxZmHpUwG087g@mail.gmail.com>

Hello Bogdan,

Thank you for looking into this!

I get warnings mostly from tm-timer. I've seen warnings from
blcore-expire, dlg-options-pinger, dlg-reinvite-pinger, dlg-timer (in
the logs, but not during my testing).
While testing, I saw only the tm-timer warnings.

I took a superficial look at the "timer_partitions" and your
explanation matches my findings. However, increasing the
"timer_partitions" makes the system unstable (doesn't matter how many
timer procs we have).
I found that I can get the most out of the system if one
"timer_partiton" is used along with one timer_proc.

With the reactor scheme, a UDP receiver can handle timer jobs, is that
right? If yes, if the UDP workers are idle, there are enough resources
to handle timer jobs, correct?

I was also increasing the TM_TABLE_ENTRIES to (1<<18) and there was a
little bit of performance increase, but I will need to test more to
come up with a valid conclusion.

On the other hand, I noticed a strange behavior on timer handling.
Take a look at:
https://github.com/OpenSIPS/opensips/issues/2797
Not sure if this is related to the warnings that I'm seeing.

The biggest performance improvement was switching to HP_MALLOC for
both pkg and shm memory.

I will keep you posted with my findings,
Ovidiu

On Thu, Mar 31, 2022 at 10:28 AM Bogdan-Andrei Iancu
<bogdan at opensips.org> wrote:
>
> Hi Ovidiu,
>
> As warnings from the timer_ticker, do you get only for the tm-utimer
> task ? I'm asking as the key question here is where the bottleneck is :
> in the whole "timer" subsystem, or in the tm-utimer task only?
>
> The TM "timer_partitions" creates multiple parallel timer lists, to
> avoid having large "amounts" of transactions handled at a moment in a
> single tm-utimer task (but rather split/partition the whole of amount of
> handled transactions into smaller chunks, to be handled one at a time in
> the timer task.
>
> The "timer_workers" creates  more than one dedicated processes for
> handling the timer tasks (so scales up the timer sub-system).
>
> If you get warnings only on tm-utimer, I suspect the bottleneck is TM
> related, mainly on performing re-transmissions (that's what that task is
> doing). So the increasing the timer-partitions should be the way to help.
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
> OpenSIPS eBootcamp 23rd May - 3rd June 2022
>    https://opensips.org/training/OpenSIPS_eBootcamp_2022/
>
> On 3/24/22 12:54 AM, Ovidiu Sas wrote:
> > Hello all,
> >
> > I'm working on tuning an opensips server. I get this pesky:
> > WARNING:core:utimer_ticker: utimer task <tm-utimer> already scheduled
> > I was trying to get rid of them by playing with the tm
> > timer_partitions parameter and the timer_workers core param.
> > By increasing any of them doesn't increase performance.
> > By increasing both of them, it actually decreases performance.
> > The server is not at limit, the load on the UDP workers is around
> > 50-60 with some spikes.
> > I have around 3500+ cps sipp traffic.
> >
> > My understanding is that by increasing the number of timer_partitions,
> > we will have more procs walking in parallel over the timer structures.
> > If we have on timer structure, we have one proc walking over it.
> > How is this working for two timer structures? What is the difference
> > between the first and the second timer structure? Should we expect
> > less work for each proc?
> >
> > For now, to reduce the occurrence of the warning log, I increased the
> > timer interval for tm-utimer from 100ms to 200ms. This should be ok as
> > the timer has the TIMER_FLAG_DELAY_ON_DELAY flag set.
> >
> > Thanks,
> > Ovidiu
> >
>


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


From denis at natden.com  Mon Mar 21 12:26:20 2022
From: denis at natden.com (Denis Alekseytsev)
Date: Mon, 21 Mar 2022 12:26:20 -0000
Subject: [OpenSIPS-Users] JSON log format
Message-ID: <d0e5986f-5a79-4e60-b806-f7be111c721f@natden.com>

Hi,

Are there any plans to introduce JSON log format and systemd-journal integration?

Thanks,
Xaled
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From eugen at teconisy.com  Wed Mar  2 23:44:17 2022
From: eugen at teconisy.com (Eugen Prieb)
Date: Wed, 02 Mar 2022 23:44:17 -0000
Subject: [OpenSIPS-Users] OpenSIPS 3.2 + MySQL - do_accounting()
In-Reply-To: <a7c3d09a-972d-0aca-8028-31606539ea24@opensips.org>
References: <98892702-b286-0ed3-9837-9391d313c36a@teconisy.com>
 <a7c3d09a-972d-0aca-8028-31606539ea24@opensips.org>
Message-ID: <8e7e32de-cca6-36b0-85db-6c678d14d923@teconisy.com>

Hello,

his is attached SIP-Trace fpr examle call. This is permanent error by 
MySQL use.

--
Mit freundlichen Grüßen,
Best regards

Eugen Prieb

Am 01.03.2022 um 15:28 schrieb Bogdan-Andrei Iancu:
> Hi Eugen,
>
> What is the capture of the call that produced that CDR ? Can you 
> reproduce it ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
> OpenSIPS eBootcamp
>   https://www.opensips.org/Training/Bootcamp
>
> On 2/23/22 11:21 PM, Eugen Prieb via Users wrote:
>> Hello,
>>
>> I will collect all CDRs, also failled, in DB.
>>
>> I see in log follow message:
>> Feb 23 21:58:33 opensips-32 /sbin/opensips[2623292]: 
>> CRITICAL:db_mysql:wrapper_single_mysql_stmt_execute: driver error 
>> (1048): Column 'to_tag' cannot be null
>>
>> As MySQL is MariaDB 10.5.12 installed. Have you any idee?
>>
>>
>
-------------- next part --------------
U 2022/03/02 23:11:52.987389 +0000 176.199.99.57:61172 -> 95.111.248.176:5060

INVITE sip:+493012030000 at 95.111.248.176 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.23:5060;branch=z9hG4bK006c3cd0eb98ec11b76f277d269fdfd3;rport
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
To: <sip:+493012030000 at 95.111.248.176>
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 5 INVITE
Contact: <sip:4901179322878 at 192.168.178.23:5060>
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 69
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:4901179322878 at 95.111.248.176>
Content-Length:   417

v=0
o=- 615152551 1 IN IP4 192.168.178.23
s=SIPPER for PhonerLite
c=IN IP4 192.168.178.23
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3874030355
a=sendrecv
        
U 2022/03/02 23:11:52.987526 +0000 95.111.248.176:5060 -> 176.199.99.57:61172

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b76f277d269fdfd3;rport=61172
To: <sip:+493012030000 at 95.111.248.176>
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 5 INVITE
Server:Class5 OpenSIPS
Content-Length: 0

        
U 2022/03/02 23:11:52.994924 +0000 95.111.248.176:5060 -> 176.199.99.57:61172

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b76f277d269fdfd3;rport=61172
To: <sip:+493012030000 at 95.111.248.176>;tag=28de.97201e0cc1f84b5f4058acdfba4ae5bf
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 5 INVITE
Proxy-Authenticate: Digest realm="95.111.248.176", nonce="Adg2o5qqx5zG1rXomjs4ijFivSdNo2tetQO2qu9/LRYA"
Server:Class5 OpenSIPS
Content-Length: 0

        
U 2022/03/02 23:11:53.531998 +0000 176.199.99.57:61172 -> 95.111.248.176:5060

INVITE sip:+493012030000 at 95.111.248.176 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.23:5060;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
To: <sip:+493012030000 at 95.111.248.176>
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 INVITE
Contact: <sip:4901179322878 at 192.168.178.23:5060>
Proxy-Authorization: Digest username="4901179322878", realm="95.111.248.176", nonce="Adg2o5qqx5zG1rXomjs4ijFivSdNo2tetQO2qu9/LRYA", uri="sip:+493012030000 at 95.111.248.176", response="f8bbe4488f0bf527d6c21038c021e434", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 69
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:4901179322878 at 95.111.248.176>
Content-Length:   417

v=0
o=- 615152551 1 IN IP4 192.168.178.23
s=SIPPER for PhonerLite
c=IN IP4 192.168.178.23
t=0 0
m=audio 5062 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:3874030355
a=sendrecv
        
U 2022/03/02 23:11:53.532054 +0000 95.111.248.176:5060 -> 176.199.99.57:61172

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport=61172
To: <sip:+493012030000 at 95.111.248.176>
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 INVITE
Server:Class5 OpenSIPS
Content-Length: 0

        
U 2022/03/02 23:11:53.882219 +0000 95.111.248.176:5060 -> 176.199.99.57:61172

SIP/2.0 100 Giving it a try
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport=61172
To: <sip:+493012030000 at 95.111.248.176>
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 INVITE
Server:Class5 OpenSIPS
Content-Length: 0

        
U 2022/03/02 23:11:53.884734 +0000 95.111.248.176:5060 -> 127.0.0.1:5080

INVITE sip:493012030000 at 127.0.0.1:5080 SIP/2.0
Record-Route: <sip:95.111.248.176;lr;ftag=2078659750;did=3de.fa0dd947>
Via: SIP/2.0/UDP 95.111.248.176:5060;branch=z9hG4bK30f3.7d39fdf6.0
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport=61172
From: "PhonerLite" <sip:493010050000 at 95.111.248.176>;tag=2078659750
To: <sip:493012030000 at 95.111.248.176>
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 INVITE
Contact: <sip:4901179322878 at 176.199.99.57:61172>
Proxy-Authorization: Digest username="4901179322878", realm="95.111.248.176", nonce="Adg2o5qqx5zG1rXomjs4ijFivSdNo2tetQO2qu9/LRYA", uri="sip:+493012030000 at 95.111.248.176", response="f8bbe4488f0bf527d6c21038c021e434", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE
Max-Forwards: 69
Supported: 100rel, replaces, from-change
User-Agent: SIPPER for PhonerLite
P-Preferred-Identity: <sip:4901179322878 at 95.111.248.176>
Content-Length: 436
P-NAT: rtpproxy_offer
P-remote-id: h323-remote-id=class4-dev
P-NextHop-IP: 5.255.95.42
Xpgk-route-retries: 0

v=0
o=- 615152551 1 IN IP4 95.111.248.176
s=SIPPER for PhonerLite
c=IN IP4 95.111.248.176
t=0 0
m=audio 44174 RTP/AVP 8 0 2 3 97 110 111 9 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:9 G722/800
U 2022/03/02 23:11:53.885374 +0000 95.111.248.176:5060 -> 127.0.0.1:5080



U 2022/03/02 23:12:03.995994 +0000 95.111.248.176:5060 -> 176.199.99.57:61172

SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP 192.168.178.23:5060;received=176.199.99.57;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport=61172
To: <sip:+493012030000 at 95.111.248.176>;tag=bcaa-16a89e664a922fe2f5878aa869f11a7f
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 INVITE
Server:Class5 OpenSIPS
Content-Length: 0

        
U 2022/03/02 23:12:04.26921 +0000 176.199.99.57:61172 -> 95.111.248.176:5060

ACK sip:+493012030000 at 95.111.248.176 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.23:5060;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
To: <sip:+493012030000 at 95.111.248.176>;tag=bcaa-16a89e664a922fe2f5878aa869f11a7f
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 ACK
Content-Length: 0

        
U 2022/03/02 23:12:04.26968 +0000 176.199.99.57:61172 -> 95.111.248.176:5060

ACK sip:+493012030000 at 95.111.248.176 SIP/2.0
Via: SIP/2.0/UDP 192.168.178.23:5060;branch=z9hG4bK006c3cd0eb98ec11b770277d269fdfd3;rport
From: "PhonerLite" <sip:4901179322878 at 95.111.248.176>;tag=2078659750
To: <sip:+493012030000 at 95.111.248.176>;tag=bcaa-16a89e664a922fe2f5878aa869f11a7f
Call-ID: 006C3CD0-EB98-EC11-B76E-277D269FDFD3 at 192.168.178.23
CSeq: 6 ACK
Content-Length: 0

        

From chester at zigbang.com  Wed Mar 30 11:45:14 2022
From: chester at zigbang.com (=?UTF-8?B?7J206riw7JuQ?=)
Date: Wed, 30 Mar 2022 11:45:14 -0000
Subject: [OpenSIPS-Users] strange INVITE transmission
In-Reply-To: <CA+-iJjZx3N+Kyk7WGzeqgq0T9Oq47h3b2p9wOdW2poEZDUCopw@mail.gmail.com>
References: <CA+-iJjYWnY44e-APV0sWN+ZedH4uTQbWqdcEBpBq01oGQ+sCug@mail.gmail.com>
 <CAN0jgiNzT3xEKfNMkYLtAEbF9Z_EMxz7D+sWWLFXr_eBWoBx0Q@mail.gmail.com>
 <CA+-iJjZ_iqDcEJqEYuwdX=-2+Kc7cr1OM4hj_qUV0C-XpawU3g@mail.gmail.com>
 <CAN0jgiOcji+2UFGuLi_3m+mxZhZhN7kSeq9d=DZVedpm6cMBig@mail.gmail.com>
 <CA+-iJjZU6Zi7jnOChHiW+BnPHxPJ=bp+pNJQAmCsJqFE0hCsKw@mail.gmail.com>
 <CAN0jgiPWt-v3f_K_vwup3D+t6-Eb3QCu9FSgK-1hpw=cjnY37Q@mail.gmail.com>
 <CA+-iJjbir-k-QKe7-iAxNHn=+Vcm6C-5+1pWNx40H8v_5yAd2w@mail.gmail.com>
 <CAN0jgiPrKc2KNHG1_J4qKbr7S1FaFvNudWCMc=TrE7NE=91fLA@mail.gmail.com>
 <CA+-iJjZx3N+Kyk7WGzeqgq0T9Oq47h3b2p9wOdW2poEZDUCopw@mail.gmail.com>
Message-ID: <CA+-iJjb5NfS_DqFxf0970ftYW5USx4idBkDzyxp26EWaH=sfbg@mail.gmail.com>

Hi guys and Daniel,

Finally, I dropped my config and generated a new config with
the "osipsconfig" command line tool including the "USE_NAT" option.
Actually my config was from one of my colleagues. He generated that config
without USE_NAT option. And I tried to make "NAT" enabled by my beginner's
hand. Some articles help me out but that causes this nightmare.
To make opensips supporting NAT, it is needed to modify many parts in the
config(script). But some articles introduce a little concept of it. I'm not
sure if that recipe worked with old versions but NOT NOW.

If you are not an expert, please use osipconfig to generate config file and
start from scratch.

Daniel helped me a lot!
Thank you

Regard
Kiwon




2022년 3월 30일 (수) 오후 12:01, 이기원 <chester at zigbang.com>님이 작성:

> Thank you for you advice, Daniel.
>
> I was testing with older versions of opensips such as 3.1 today.
>
> You give me wonderful information. I'll spend some time with opensip and
> share my results with you.
>
> Good night
>
>
>
> 2022년 3월 30일 (수) 오전 11:50, Daniel Zanutti <daniel.zanutti at gmail.com>님이 작성:
>
>> GMT -3:00 here
>>
>> I'm sending a copy of a config we have here, for Opensips 1.11.11.
>>
>> I configured some months ago to a developer use on his machine and test
>> with no DB backend configured. This config will allow any user/pass and
>> should handle NAT seamlessly. Please take a look and study how it works.
>> You could install opensips 1.11.11 and test it or port the config to 3.x,
>> we use 1.11.11 in production with no known bugs.
>>
>> Don't forget that you are handling NAT using SIP, to also handle NAT for
>> audio (RTP), you're gonna need RTPPROXY. It's already configured on this
>> script, if you don't want it, just comment the module and all errors you
>> find.
>>
>> Hope this helps.
>>
>> On Tue, Mar 29, 2022 at 10:23 PM 이기원 <chester at zigbang.com> wrote:
>>
>>> Hi Daniel,
>>>
>>> I see the empty value for the received column.
>>>
>>> [image: image.png]
>>>
>>> However I read the right IP address for that device. Opensips resolves
>>> its IP address but doesn't write it down in DB.
>>>
>>> Any options?
>>>
>>> By the way, what timezone do you live in? I'm in GMT+9 Seoul, Korea.
>>>
>>> Thank you
>>>
>>> Kiwon
>>>
>>> 2022년 3월 29일 (화) 오후 11:41, Daniel Zanutti <daniel.zanutti at gmail.com>님이
>>> 작성:
>>>
>>>> Hi Kiwon
>>>>
>>>> You are almost there.
>>>>
>>>> On location table, the Contact column will be the same as the SIP
>>>> Contact header. The public IP will be on the "received" column. Take a look
>>>> at a production server we have here:
>>>>
>>>> [image: image.png]
>>>>
>>>> Could you check this is correct on your table?
>>>>
>>>> This e-mail is a Gmail account, add me for chat if you want.
>>>>
>>>>
>>>>
>>>> On Tue, Mar 29, 2022 at 4:43 AM 이기원 <chester at zigbang.com> wrote:
>>>>
>>>>> Hi Daniel,
>>>>>
>>>>> Thank you for your kind help again.
>>>>>
>>>>> In my first trial, I told you there was no response from opensips. Now
>>>>> I realized that it was because of crashing due to "Mar 29 14:38:36 [70]
>>>>> ERROR:core:check_actions: check failed for function <nat_uac_test>,
>>>>> /etc/opensips/opensips.cfg:256
>>>>> Mar 29 14:38:36 [70] ERROR:core:main: bad function call in config
>>>>> file".
>>>>> That's why there is no response from opensips.
>>>>> nat_uac_test's parameter should be number. :)
>>>>>
>>>>> And I revised the config file as your second advice, but it's just the
>>>>> same as before I put the NAT helper script. Please take a look at the call
>>>>> flow.
>>>>> [image: image.png]
>>>>> In the picture,
>>>>> * 175.223.34.31 is "zoiper" that is a soft-phone on iPhone
>>>>> * 10.0.0.177 is opensips server in AWS. Its external IP address
>>>>> is 52.79.209.75 and I put "advertised_address" with that IP address in cfg
>>>>> file (
>>>>> https://blog.opensips.org/2017/10/25/running-opensips-in-the-cloud/ )
>>>>> * 192.168.10.187 is testing phone. Its public IP address is
>>>>> 14.52.242.236. The picture below is a registration flow for the testing
>>>>> phone.
>>>>> [image: image.png]
>>>>>
>>>>>
>>>>> and some logs from opensips------------------------------
>>>>>
>>>>> Mar 29 14:50:48 ip-10-0-0-177 /usr/sbin/opensips[45]: Fix contact -
>>>>> M=REGISTER RURI=sip:52.79.209.75:5090 F=sip:6002 at 52.79.209.75:5090 T=
>>>>> sip:6002 at 52.79.209.75:5090 IP=14.52.252.236 ID=
>>>>> 147053064617476-337321534728858 at 192.168.10.187
>>>>> Mar 29 14:50:48 ip-10-0-0-177 /usr/sbin/opensips[48]: Fix contact -
>>>>> M=REGISTER RURI=sip:52.79.209.75:5090 F=sip:6002 at 52.79.209.75:5090 T=
>>>>> sip:6002 at 52.79.209.75:5090 IP=14.52.252.236 ID=
>>>>> 147053064617476-337321534728858 at 192.168.10.187
>>>>> Mar 29 14:51:04 ip-10-0-0-177 /usr/sbin/opensips[45]: Fix contact -
>>>>> M=INVITE RURI=sip:6002 at 52.79.209.75:5090;transport=UDP
>>>>> F=sip:6001 at 52.79.209.75:5090;transport=UDP T=
>>>>> sip:6002 at 52.79.209.75:5090 IP=175.223.34.31
>>>>> ID=hqy06Z2nJ2lT0z1h4DKKUg..
>>>>> Mar 29 14:51:04 ip-10-0-0-177 /usr/sbin/opensips[48]: Fix contact -
>>>>> M=INVITE RURI=sip:6002 at 52.79.209.75:5090;transport=UDP
>>>>> F=sip:6001 at 52.79.209.75:5090;transport=UDP T=
>>>>> sip:6002 at 52.79.209.75:5090 IP=175.223.34.31
>>>>> ID=hqy06Z2nJ2lT0z1h4DKKUg..
>>>>>
>>>>>
>>>>>
>>>>> The document says
>>>>> 1.5.4.  fix_nated_register()
>>>>>
>>>>> The function creates a URI consisting of the source IP, port, and
>>>>> protocol and stores the URI in an Attribute-Value-Pair. The URI will be
>>>>> appended as "received" parameter to Contact in 200 OK and registrar will
>>>>> store it in the user location database.
>>>>> So I looked into mysql, especially the location table. I read that the
>>>>> contact address is still written with a private IP address.
>>>>>
>>>>> [image: image.png]
>>>>>
>>>>> Hmm. What am I missing?
>>>>>
>>>>> Thank you again.
>>>>>
>>>>> Regards
>>>>> Kiwon
>>>>>
>>>>>
>>>>>
>>>>> 2022년 3월 29일 (화) 오후 1:08, Daniel Zanutti <daniel.zanutti at gmail.com>님이
>>>>> 작성:
>>>>>
>>>>>> Hi Kiwon
>>>>>>
>>>>>> You applied at beginning, before loose_route. Not sure it gonna work
>>>>>> this way, i sent you exactly line where to put the code.
>>>>>> Move this:
>>>>>>  if (nat_uac_test(23)) {
>>>>>> if (is_method("REGISTER")) {
>>>>>> fix_nated_register();
>>>>>> setbflag("NAT");
>>>>>> } else {
>>>>>> fix_nated_contact();
>>>>>> setflag("NAT");
>>>>>> }
>>>>>> }
>>>>>>
>>>>>> After:
>>>>>> t_check_trans();
>>>>>>
>>>>>> Anyway, I suggest you add some log to confirm messages are coming.
>>>>>> Put this line right after main route:
>>>>>> xlog("L_ERR","MESSAGE RECEIVED $rm [$fu/$tu/$ru/$ci/$si]");
>>>>>>
>>>>>> There's no big deal on Opensips, everything comes to main route,
>>>>>> internal transaction responses comes from specific routes. (
>>>>>> https://www.opensips.org/Documentation/Script-Routes-3-1)
>>>>>>
>>>>>> I have some spare time tomorrow, send me a direct message if you need
>>>>>> help.
>>>>>>
>>>>>>
>>>>>> On Tue, Mar 29, 2022 at 12:43 AM 이기원 <chester at zigbang.com> wrote:
>>>>>>
>>>>>>> Hi Daniel and opensips users goup,
>>>>>>> Thank you for answering my questions.
>>>>>>>
>>>>>>> I missed to inform you about the opensips version I'm trying. It is
>>>>>>> 3.2.5.
>>>>>>>
>>>>>>> After modifying config, opensips does not respond for the REGISTER
>>>>>>> requests from phones.
>>>>>>> Actually I already tried the following
>>>>>>> https://kb.smartvox.co.uk/opensips/nat-contact-and-via-fixing-in-sip-part-3/
>>>>>>> article but I have the same problem - no response for REGISTERs.
>>>>>>>
>>>>>>> Is there any way to know why opensips ignores or does not respond
>>>>>>> for REGISTERs?
>>>>>>> Please find my new opensips.cfg that Diniel's advice is applied.
>>>>>>>
>>>>>>>
>>>>>>> Thank you
>>>>>>>
>>>>>>> Regards
>>>>>>> Kiwon
>>>>>>>
>>>>>>> 2022년 3월 28일 (월) 오후 10:47, Daniel Zanutti <daniel.zanutti at gmail.com>님이
>>>>>>> 작성:
>>>>>>>
>>>>>>>> Hi Kiwon
>>>>>>>>
>>>>>>>> You need to handle NAT scenarios. Try putting this code on line
>>>>>>>> 254, right after "t_check_trans()":
>>>>>>>>
>>>>>>>> if (nat_uac_test("7"))
>>>>>>>> {
>>>>>>>>   #nathelper
>>>>>>>>   if(is_method("REGISTER"))
>>>>>>>>     fix_nated_register();
>>>>>>>>   else
>>>>>>>>     fix_nated_contact();
>>>>>>>>   xlog("L_NOTICE", "Fix contact - M=$rm RURI=$ru F=$fu T=$tu IP=$si
>>>>>>>> ID=$ci\n");
>>>>>>>> }
>>>>>>>>
>>>>>>>> You also need to enable nathelper module. The reason is that you
>>>>>>>> need to use the public IP/Port that sent data to opensips and ignore the
>>>>>>>> Contact.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Mon, Mar 28, 2022 at 5:20 AM 이기원 <chester at zigbang.com> wrote:
>>>>>>>>
>>>>>>>>> Hi guys, I'm trying to introduce opensips into my company these
>>>>>>>>> days.
>>>>>>>>>
>>>>>>>>> I'm testing with many other phones. During the test, I had a very
>>>>>>>>> strange symptom.
>>>>>>>>>
>>>>>>>>> Please, take look at the picture below first. (Please find my
>>>>>>>>> screenshot from the link below)
>>>>>>>>> A. The most left one 14.52.252.236 is a phone (with hardware).
>>>>>>>>> B. 10.0.0.177 is proxy which is woking on cloud and its external
>>>>>>>>> IP is 58.79.209.75
>>>>>>>>> C. 175.223.34.31 is a zoiper softphone which is working on iphone
>>>>>>>>> D. 192.168.10.187 is the private IP address of A - a phone which
>>>>>>>>> is the most left one (14.52.252.236)
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> https://drive.google.com/file/d/14zAREWLsluIa1TcU7tZJLSgff-iPqBDA/view?usp=sharing
>>>>>>>>>
>>>>>>>>> As you can see, C is calling A but opensips transmits the INVITE
>>>>>>>>> rqeuset to A's private IP address.
>>>>>>>>> INVITE request should be transmitted A's public IP address - In
>>>>>>>>> this scenario 14.52.252.236 (A)
>>>>>>>>>
>>>>>>>>> I also attach my opensips.cfg. What's wrong with me?
>>>>>>>>>
>>>>>>>>> Thank you
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>> Kiwon
>>>>>>>>> _______________________________________________
>>>>>>>>> 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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