From spanda at 3clogic.com  Mon Jul  1 09:08:51 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 1 Jul 2024 14:38:51 +0530
Subject: [OpenSIPS-Users] I need help on opensips-trap and opensips-dbg
	package for debugging .
Message-ID: <CALf-Q46=9_duNDpxW0YY7e4gHmS2Qgxszyx7zRq2tbO8CnjpVA@mail.gmail.com>

Hi All ,

I am using the below  linux version and amd base architecture .

[ opensips-codechanged-3.2]# cat /etc/*release
Amazon Linux release 2023.4.20240401 (Amazon Linux)
NAME="Amazon Linux"
VERSION="2023"
ID="amzn"
ID_LIKE="fedora"
VERSION_ID="2023"
PLATFORM_ID="platform:al2023"
PRETTY_NAME="Amazon Linux 2023.4.20240401"
ANSI_COLOR="0;33"
CPE_NAME="cpe:2.3:o:amazon:amazon_linux:2023"
HOME_URL="https://aws.amazon.com/linux/amazon-linux-2023/"
DOCUMENTATION_URL="https://docs.aws.amazon.com/linux/"
SUPPORT_URL="https://aws.amazon.com/premiumsupport/"
BUG_REPORT_URL="https://github.com/amazonlinux/amazon-linux-2023"
VENDOR_NAME="AWS"
VENDOR_URL="https://aws.amazon.com/"
SUPPORT_END="2028-03-15"
Amazon Linux release 2023.4.20240401 (Amazon Linux)
[ opensips-codechanged-3.2]#
[ opensips-codechanged-3.2]# uname -r
6.1.82-99.168.amzn2023.x86_64
[ opensips-codechanged-3.2]#

I am getting timer waring while starting opensips which I have posted in
the forum earlier as well . But not getting any proper solution for this .
Now this is becoming critical for me . I wanted to take core file with
backtrace also earlier I got suggestion to take opensips trap command . But
opensips-cli says no trap module loaded and I am also not able to install
opensips-dbg package on this system .

Is there any proper guideline to install opensips-cli with trap module and
opensips-dbg package on the above linux . If not then what is the best
suitable version of linux I must use [for opensips where I can install them
easily . Please do suggest .


Attached the installation doc which I used to follow to install opensips
and opensips-cli manually . Please suggest to me what I should do here .



*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From alain.bieuzent at free.fr  Tue Jul  2 10:14:09 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Tue, 02 Jul 2024 12:14:09 +0200
Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and
 3.2.19 Minor Releases
In-Reply-To: <bf5602ab-32e1-4193-b420-fd9cadda15da@opensips.org>
References: <98729b64-72b2-726d-1532-31a89b0522bb@opensips.org>
 <bf5602ab-32e1-4193-b420-fd9cadda15da@opensips.org>
Message-ID: <0C79A4D3-9FCF-4A57-B185-2A2ECC6CD5DE@free.fr>

Hi liviu,

 

any idea when the repository will be updated to the latest versions? (apt.opensips.org)

 

thanks

 

De : Users <users-bounces at lists.opensips.org> au nom de Liviu Chircu <liviu at opensips.org>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mercredi 19 juin 2024 à 15:06
À : OpenSIPS Users <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases

 

Hello,

A new round of stable minor releases is now out: 3.4.6 and 3.2.19.  Note that support for 3.2 LTS release has ended, so make sure to upgrade to 3.4 LTS in order to continue receiving fixes.

Finally, a first stable release candidate for 3.5 branch is now out: 3.5.0-rc1

Full changelogs:

https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog
https://opensips.org/pub/opensips/3.4.6/ChangeLog
https://opensips.org/pub/opensips/3.2.19/ChangeLog

Please enjoy!
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com
On 13.06.2024 17:05, Liviu Chircu wrote:

Hi all,

The 3.4.6 and 3.2.19 OpenSIPS minor versions are scheduled for release on Wednesday, June 19th.  Please note that this will mark the end-of-life for the 3.2 LTS version, according to the OpenSIPS release policy.

Moreover, as the beta testing for release 3.5 is still ongoing, we will mark the current progress with a new release candidate: 3.5.0-rc1

In preparation for the releases, we impose the usual freeze on any significant fixes (as complexity) on the stable branches, in order to ensure a safe window for testing in the days ahead.

Finally, please make sure to ping any outstanding issues on the GitHub issue tracker that may have skipped our attention -- thank you in advance!

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


_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________ 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  Wed Jul  3 10:00:35 2024
From: vinayak.makwana at ecosmob.com (Vinayak Makwana)
Date: Wed, 3 Jul 2024 15:30:35 +0530
Subject: [OpenSIPS-Users] Handle wss to udp Call with rtp_relay module
Message-ID: <CAPHmyfz-XcRjSYdfRg8dzjoxZ1jN_aT5LVg_qa0BYANkeHpujQ@mail.gmail.com>

Hello all,
             I am using opensips 3.4.6 version and in order to handle rtp i
have configured rtpengine and rtp_relay module in opensips script.
For the UDP to UDP call I am using the same rtp_relay and rtp_relay_peer
variable and it's working fine. but when i configured for wss to udp call
at time not able to get audio both sides.

Here's my sample code.

route[INIT_INVITE]{
              if (isflagset("SRC_WS") && !isbflagset("DST_WS")) {
               xlog("L_INFO", " [$ci] [$rm] [WSS to UDP]");
               $rtp_relay="RTP/AVP replace-session-connection
replace-origin ICE=remove";
               $rtp_relay_peer="UDP/TLS/RTP/SAVPF trust-address ICE=force
rtcp-mux-offer SDES-off replace-session-connection replace-origin
generate-mid";
               rtp_relay_engage("rtpengine");
}

Let me know if I have missed anything or any changes required.


Thanks
Vinayak

-- 
* <https://www.ecosmob.com/agile-testing-webinar/>
Disclaimer*

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From venefax at gmail.com  Wed Jul  3 13:01:02 2024
From: venefax at gmail.com (Saint Michael)
Date: Wed, 3 Jul 2024 09:01:02 -0400
Subject: [OpenSIPS-Users] Originating address in multihomed computer
Message-ID: <CAC9cSOCYD_2rh0-ahuZ55-_XXyHKF=LDgU_jJR5BGCfECD4ddA@mail.gmail.com>

I have an opensips box with 40+ IP addresses. No matter the call's
"received at" address, I need the second leg to use the same IP, so
when the carrier receives it, he sees the same IP as "source" that I
saw as "received at".
How can I do this? is it possible?
Philip


From venefax at gmail.com  Wed Jul  3 21:07:53 2024
From: venefax at gmail.com (Saint Michael)
Date: Wed, 3 Jul 2024 17:07:53 -0400
Subject: [OpenSIPS-Users] Is there any way to include files in the
	configuration?
Message-ID: <CAC9cSOBnKqyj8fpT+RyaonAx3D3+apsCqwxWh5SRMcMCE5K1bA@mail.gmail.com>

I need something like
#include "/etc/opensips/test.cfg", inside opensips.cfg
Is this possible using some mechanism?
Philip


From brett at nemeroff.com  Wed Jul  3 23:05:10 2024
From: brett at nemeroff.com (Brett Nemeroff)
Date: Wed, 3 Jul 2024 18:05:10 -0500
Subject: [OpenSIPS-Users] Is there any way to include files in the
	configuration?
In-Reply-To: <CAC9cSOBnKqyj8fpT+RyaonAx3D3+apsCqwxWh5SRMcMCE5K1bA@mail.gmail.com>
References: <CAC9cSOBnKqyj8fpT+RyaonAx3D3+apsCqwxWh5SRMcMCE5K1bA@mail.gmail.com>
Message-ID: <CAPwC5wwf_pa9qAH3aPi0+FXEzLYM79P3dTFfrpaUpaPz+mbgXw@mail.gmail.com>

OpenSIPS depends on external preprocessors for this kind of functionality.
See if this peg helps you out. If not, let us know:

https://www.opensips.org/Documentation/Templating-Config-Files-3-2

-Brett


On Wed, Jul 3, 2024 at 4:09 PM Saint Michael <venefax at gmail.com> wrote:

> I need something like
> #include "/etc/opensips/test.cfg", inside opensips.cfg
> Is this possible using some mechanism?
> Philip
>
> _______________________________________________
> 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  Wed Jul  3 23:24:17 2024
From: venefax at gmail.com (Saint Michael)
Date: Wed, 3 Jul 2024 19:24:17 -0400
Subject: [OpenSIPS-Users] Is there any way to include files in the
	configuration?
In-Reply-To: <CAPwC5wwf_pa9qAH3aPi0+FXEzLYM79P3dTFfrpaUpaPz+mbgXw@mail.gmail.com>
References: <CAC9cSOBnKqyj8fpT+RyaonAx3D3+apsCqwxWh5SRMcMCE5K1bA@mail.gmail.com>
 <CAPwC5wwf_pa9qAH3aPi0+FXEzLYM79P3dTFfrpaUpaPz+mbgXw@mail.gmail.com>
Message-ID: <CAC9cSODgRdvjtz0MtW=AzFySThYpn5n80BKz2MxrPA=JGkbM=Q@mail.gmail.com>

Include_file "/etc/opensips/demo.cfg"

On Wed, Jul 3, 2024, 7:10 PM Brett Nemeroff <brett at nemeroff.com> wrote:

> OpenSIPS depends on external preprocessors for this kind of functionality.
> See if this peg helps you out. If not, let us know:
>
> https://www.opensips.org/Documentation/Templating-Config-Files-3-2
>
> -Brett
>
>
> On Wed, Jul 3, 2024 at 4:09 PM Saint Michael <venefax at gmail.com> wrote:
>
>> I need something like
>> #include "/etc/opensips/test.cfg", inside opensips.cfg
>> Is this possible using some mechanism?
>> Philip
>>
>> _______________________________________________
>> 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 alain.bieuzent at free.fr  Thu Jul  4 08:35:14 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Thu, 04 Jul 2024 10:35:14 +0200
Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and
 3.2.19 Minor Releases
In-Reply-To: <0C79A4D3-9FCF-4A57-B185-2A2ECC6CD5DE@free.fr>
References: <98729b64-72b2-726d-1532-31a89b0522bb@opensips.org>
 <bf5602ab-32e1-4193-b420-fd9cadda15da@opensips.org>
 <0C79A4D3-9FCF-4A57-B185-2A2ECC6CD5DE@free.fr>
Message-ID: <3B183389-768F-4E3B-B513-88A9F77B953B@free.fr>

Hi,

 

Any update on this ?

I’m running Debian bullseye, when i search in the repo, the candidate version is 3.4.5

 

apt-cache policy opensips

opensips:

  InstallĂŠÂ : 3.4.4-1

  Candidat : 3.4.5-1

 Table de version :

     3.4.5-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

 *** 3.4.4-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

        100 /var/lib/dpkg/status

     3.4.3-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

     3.4.2-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

     3.4.1-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

     3.4.0-rc1-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

     3.4.0-beta-1 500

        500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages

 

Regards

 

De : Users <users-bounces at lists.opensips.org> au nom de Alain Bieuzent <alain.bieuzent at free.fr>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 2 juillet 2024 à 12:17
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases

 

Hi liviu,

 

any idea when the repository will be updated to the latest versions? (apt.opensips.org)

 

thanks

 

De : Users <users-bounces at lists.opensips.org> au nom de Liviu Chircu <liviu at opensips.org>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mercredi 19 juin 2024 à 15:06
À : OpenSIPS Users <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases

 

Hello,

A new round of stable minor releases is now out: 3.4.6 and 3.2.19.  Note that support for 3.2 LTS release has ended, so make sure to upgrade to 3.4 LTS in order to continue receiving fixes.

Finally, a first stable release candidate for 3.5 branch is now out: 3.5.0-rc1

Full changelogs:

https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog
https://opensips.org/pub/opensips/3.4.6/ChangeLog
https://opensips.org/pub/opensips/3.2.19/ChangeLog

Please enjoy!
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com
On 13.06.2024 17:05, Liviu Chircu wrote:

Hi all,

The 3.4.6 and 3.2.19 OpenSIPS minor versions are scheduled for release on Wednesday, June 19th.  Please note that this will mark the end-of-life for the 3.2 LTS version, according to the OpenSIPS release policy.

Moreover, as the beta testing for release 3.5 is still ongoing, we will mark the current progress with a new release candidate: 3.5.0-rc1

In preparation for the releases, we impose the usual freeze on any significant fixes (as complexity) on the stable branches, in order to ensure a safe window for testing in the days ahead.

Finally, please make sure to ping any outstanding issues on the GitHub issue tracker that may have skipped our attention -- thank you in advance!

Happy testing,
-- 
Liviu Chircu
www.twitter.com/liviuchircu | www.opensips-solutions.com
 
_______________________________________________
Users mailing list
Users at lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
_______________________________________________ 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 razvan at opensips.org  Thu Jul  4 08:47:58 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Thu, 4 Jul 2024 11:47:58 +0300
Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and
 3.2.19 Minor Releases
In-Reply-To: <3B183389-768F-4E3B-B513-88A9F77B953B@free.fr>
References: <98729b64-72b2-726d-1532-31a89b0522bb@opensips.org>
 <bf5602ab-32e1-4193-b420-fd9cadda15da@opensips.org>
 <0C79A4D3-9FCF-4A57-B185-2A2ECC6CD5DE@free.fr>
 <3B183389-768F-4E3B-B513-88A9F77B953B@free.fr>
Message-ID: <de926cfc-089d-42ad-9b5b-ab0739a70570@opensips.org>

Hi, Alain!

It seems like the apt repository got stuck - I've just resumed it now 
and rebuilt the packages - they should be ready in a couple of hours.

Best regards,

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

On 7/4/24 11:35 AM, Alain Bieuzent wrote:
> Hi,
> 
>   
> 
> Any update on this ?
> 
> I’m running Debian bullseye, when i search in the repo, the candidate version is 3.4.5
> 
>   
> 
> apt-cache policy opensips
> 
> opensips:
> 
>    InstallĂŠÂ : 3.4.4-1
> 
>    Candidat : 3.4.5-1
> 
>   Table de version :
> 
>       3.4.5-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>   *** 3.4.4-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>          100 /var/lib/dpkg/status
> 
>       3.4.3-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>       3.4.2-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>       3.4.1-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>       3.4.0-rc1-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>       3.4.0-beta-1 500
> 
>          500 https://apt.opensips.org bullseye/3.4-releases amd64 Packages
> 
>   
> 
> Regards
> 
>   
> 
> De : Users <users-bounces at lists.opensips.org> au nom de Alain Bieuzent <alain.bieuzent at free.fr>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
> Date : mardi 2 juillet 2024 à 12:17
> À : OpenSIPS users mailling list <users at lists.opensips.org>
> Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases
> 
>   
> 
> Hi liviu,
> 
>   
> 
> any idea when the repository will be updated to the latest versions? (apt.opensips.org)
> 
>   
> 
> thanks
> 
>   
> 
> De : Users <users-bounces at lists.opensips.org> au nom de Liviu Chircu <liviu at opensips.org>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
> Date : mercredi 19 juin 2024 à 15:06
> À : OpenSIPS Users <users at lists.opensips.org>
> Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases
> 
>   
> 
> Hello,
> 
> A new round of stable minor releases is now out: 3.4.6 and 3.2.19.  Note that support for 3.2 LTS release has ended, so make sure to upgrade to 3.4 LTS in order to continue receiving fixes.
> 
> Finally, a first stable release candidate for 3.5 branch is now out: 3.5.0-rc1
> 
> Full changelogs:
> 
> https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog
> https://opensips.org/pub/opensips/3.4.6/ChangeLog
> https://opensips.org/pub/opensips/3.2.19/ChangeLog
> 
> Please enjoy!
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> On 13.06.2024 17:05, Liviu Chircu wrote:
> 
> Hi all,
> 
> The 3.4.6 and 3.2.19 OpenSIPS minor versions are scheduled for release on Wednesday, June 19th.  Please note that this will mark the end-of-life for the 3.2 LTS version, according to the OpenSIPS release policy.
> 
> Moreover, as the beta testing for release 3.5 is still ongoing, we will mark the current progress with a new release candidate: 3.5.0-rc1
> 
> In preparation for the releases, we impose the usual freeze on any significant fixes (as complexity) on the stable branches, in order to ensure a safe window for testing in the days ahead.
> 
> Finally, please make sure to ping any outstanding issues on the GitHub issue tracker that may have skipped our attention -- thank you in advance!
> 
> Happy testing,
> 
> 
> Hi,
> 
> Any update on this ?
> 
> I’m running Debian bullseye, when i search in the repo, the candidate 
> version is 3.4.5
> 
> apt-cache policy opensips
> 
> opensips:
> 
> InstallÊ : 3.4.4-1
> 
>    Candidat : 3.4.5-1
> 
> Table de version :
> 
> 3.4.5-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
> *** 3.4.4-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
>          100 /var/lib/dpkg/status
> 
>       3.4.3-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
>       3.4.2-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
>       3.4.1-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
>       3.4.0-rc1-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
>       3.4.0-beta-1 500
> 
>          500 https://apt.opensips.org 
> <https://apt.opensips.org>bullseye/3.4-releases amd64 Packages
> 
> Regards
> 
> *De : *Users <users-bounces at lists.opensips.org> au nom de Alain Bieuzent 
> <alain.bieuzent at free.fr>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *mardi 2 juillet 2024 à 12:17
> *À : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Objet : *Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 
> and 3.2.19 Minor Releases
> 
> Hi liviu,
> 
> any idea when the repository will be updated to the latest versions? 
> (apt.opensips.org)
> 
> thanks
> 
> *De : *Users <users-bounces at lists.opensips.org> au nom de Liviu Chircu 
> <liviu at opensips.org>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *mercredi 19 juin 2024 à 15:06
> *À : *OpenSIPS Users <users at lists.opensips.org>
> *Objet : *Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 
> and 3.2.19 Minor Releases
> 
> Hello,
> 
> A new round of stable minor releases is now out: *3.4.6 *and *3.2.19*.  
> Note that support for *3.2 LTS *release has ended, so make sure to 
> upgrade to *3.4* *LTS* in order to continue receiving fixes.
> 
> Finally, a first stable release candidate for *3.5* branch is now out: 
> *3.5.0-rc1*
> 
> Full changelogs:
> 
> https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog 
> <https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog>
> https://opensips.org/pub/opensips/3.4.6/ChangeLog 
> <https://opensips.org/pub/opensips/3.4.6/ChangeLog>
> https://opensips.org/pub/opensips/3.2.19/ChangeLog 
> <https://opensips.org/pub/opensips/3.2.19/ChangeLog>
> 
> Please enjoy!
> 
> Liviu Chircu
> 
> www.twitter.com/liviuchircu  <http://www.twitter.com/liviuchircu>  |www.opensips-solutions.com  <http://www.opensips-solutions.com>
> 
> On 13.06.2024 17:05, Liviu Chircu wrote:
> 
>     Hi all,
> 
>     The *3.4.6* and *3.2.19* OpenSIPS minor versions are scheduled for
>     release on *Wednesday, June 19th*.  Please note that this will mark
>     the *end-of-life* for the *3.2* *LTS *version, according to the
>     OpenSIPS release policy
>     <https://www.opensips.org/Development/Development>.
> 
>     Moreover, as the beta testing for release *3.5* is still ongoing, we
>     will mark the current progress with a new release candidate: *3.5.0-rc1*
> 
>     In preparation for the releases, we impose the usual freeze on any
>     significant fixes (as complexity) on the stable branches, in order
>     to ensure a safe window for testing in the days ahead.
> 
>     Finally, please make sure to ping any outstanding issues on the
>     GitHub issue tracker that may have skipped our attention -- /thank
>     you/ in advance!
> 
>     Happy testing,
> 
>     -- 
> 
>     Liviu Chircu
> 
>     www.twitter.com/liviuchircu  <http://www.twitter.com/liviuchircu>  |www.opensips-solutions.com  <http://www.opensips-solutions.com>
> 
>     _______________________________________________
> 
>     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
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From slackway2me at gmail.com  Thu Jul  4 14:13:13 2024
From: slackway2me at gmail.com (Alexey)
Date: Thu, 4 Jul 2024 19:13:13 +0500
Subject: [OpenSIPS-Users] Originating address in multihomed computer
In-Reply-To: <CAC9cSOCYD_2rh0-ahuZ55-_XXyHKF=LDgU_jJR5BGCfECD4ddA@mail.gmail.com>
References: <CAC9cSOCYD_2rh0-ahuZ55-_XXyHKF=LDgU_jJR5BGCfECD4ddA@mail.gmail.com>
Message-ID: <CAPc6LUHWqHu=iLH_mjnn5syD=HtrNhg0AWz3tF_rVp7xVxwUJQ@mail.gmail.com>

I think you should start from $socket_in [1] and $socket_out [2] variables.

[1] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc80
[2] https://www.opensips.org/Documentation/Script-CoreVar-3-5#toc81

-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From alain.bieuzent at free.fr  Thu Jul  4 15:35:31 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Thu, 04 Jul 2024 17:35:31 +0200
Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and
 3.2.19 Minor Releases
In-Reply-To: <de926cfc-089d-42ad-9b5b-ab0739a70570@opensips.org>
References: <98729b64-72b2-726d-1532-31a89b0522bb@opensips.org>
 <bf5602ab-32e1-4193-b420-fd9cadda15da@opensips.org>
 <0C79A4D3-9FCF-4A57-B185-2A2ECC6CD5DE@free.fr>
 <3B183389-768F-4E3B-B513-88A9F77B953B@free.fr>
 <de926cfc-089d-42ad-9b5b-ab0739a70570@opensips.org>
Message-ID: <26CDDFB7-935C-479E-8266-1F468D052CCE@free.fr>

Hi Răzvan !

It works now, thanks

Le 04/07/2024 10:50, « Users au nom de Răzvan Crainea » <users-bounces at lists.opensips.org <mailto:users-bounces at lists.opensips.org> au nom de razvan at opensips.org <mailto:razvan at opensips.org>> a écrit :


Hi, Alain!


It seems like the apt repository got stuck - I've just resumed it now 
and rebuilt the packages - they should be ready in a couple of hours.


Best regards,


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


On 7/4/24 11:35 AM, Alain Bieuzent wrote:
> Hi,
> 
> 
> 
> Any update on this ?
> 
> I’m running Debian bullseye, when i search in the repo, the candidate version is 3.4.5
> 
> 
> 
> apt-cache policy opensips
> 
> opensips:
> 
> InstallĂŠÂ : 3.4.4-1
> 
> Candidat : 3.4.5-1
> 
> Table de version :
> 
> 3.4.5-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> *** 3.4.4-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 100 /var/lib/dpkg/status
> 
> 3.4.3-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 3.4.2-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 3.4.1-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 3.4.0-rc1-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 3.4.0-beta-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> bullseye/3.4-releases amd64 Packages
> 
> 
> 
> Regards
> 
> 
> 
> De : Users <users-bounces at lists.opensips.org <mailto:users-bounces at lists.opensips.org>> au nom de Alain Bieuzent <alain.bieuzent at free.fr <mailto:alain.bieuzent at free.fr>>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> Date : mardi 2 juillet 2024 à 12:17
> À : OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases
> 
> 
> 
> Hi liviu,
> 
> 
> 
> any idea when the repository will be updated to the latest versions? (apt.opensips.org)
> 
> 
> 
> thanks
> 
> 
> 
> De : Users <users-bounces at lists.opensips.org <mailto:users-bounces at lists.opensips.org>> au nom de Liviu Chircu <liviu at opensips.org <mailto:liviu at opensips.org>>
> Répondre à : OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> Date : mercredi 19 juin 2024 à 15:06
> À : OpenSIPS Users <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> Objet : Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 and 3.2.19 Minor Releases
> 
> 
> 
> Hello,
> 
> A new round of stable minor releases is now out: 3.4.6 and 3.2.19. Note that support for 3.2 LTS release has ended, so make sure to upgrade to 3.4 LTS in order to continue receiving fixes.
> 
> Finally, a first stable release candidate for 3.5 branch is now out: 3.5.0-rc1
> 
> Full changelogs:
> 
> https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog <https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog>
> https://opensips.org/pub/opensips/3.4.6/ChangeLog <https://opensips.org/pub/opensips/3.4.6/ChangeLog>
> https://opensips.org/pub/opensips/3.2.19/ChangeLog <https://opensips.org/pub/opensips/3.2.19/ChangeLog>
> 
> Please enjoy!
> Liviu Chircu
> www.twitter.com/liviuchircu | www.opensips-solutions.com
> On 13.06.2024 17:05, Liviu Chircu wrote:
> 
> Hi all,
> 
> The 3.4.6 and 3.2.19 OpenSIPS minor versions are scheduled for release on Wednesday, June 19th. Please note that this will mark the end-of-life for the 3.2 LTS version, according to the OpenSIPS release policy.
> 
> Moreover, as the beta testing for release 3.5 is still ongoing, we will mark the current progress with a new release candidate: 3.5.0-rc1
> 
> In preparation for the releases, we impose the usual freeze on any significant fixes (as complexity) on the stable branches, in order to ensure a safe window for testing in the days ahead.
> 
> Finally, please make sure to ping any outstanding issues on the GitHub issue tracker that may have skipped our attention -- thank you in advance!
> 
> Happy testing,
> 
> 
> Hi,
> 
> Any update on this ?
> 
> I’m running Debian bullseye, when i search in the repo, the candidate 
> version is 3.4.5
> 
> apt-cache policy opensips
> 
> opensips:
> 
> InstallĂŠÂ : 3.4.4-1
> 
> Candidat : 3.4.5-1
> 
> Table de version :
> 
> 3.4.5-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> *** 3.4.4-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> 100 /var/lib/dpkg/status
> 
> 3.4.3-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> 3.4.2-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> 3.4.1-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> 3.4.0-rc1-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> 3.4.0-beta-1 500
> 
> 500 https://apt.opensips.org <https://apt.opensips.org> 
> <https://apt.opensips.org>bullseye/3.4-releases <https://apt.opensips.org&gt;bullseye/3.4-releases> amd64 Packages
> 
> Regards
> 
> *De : *Users <users-bounces at lists.opensips.org <mailto:users-bounces at lists.opensips.org>> au nom de Alain Bieuzent 
> <alain.bieuzent at free.fr <mailto:alain.bieuzent at free.fr>>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> *Date : *mardi 2 juillet 2024 à 12:17
> *À : *OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> *Objet : *Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 
> and 3.2.19 Minor Releases
> 
> Hi liviu,
> 
> any idea when the repository will be updated to the latest versions? 
> (apt.opensips.org)
> 
> thanks
> 
> *De : *Users <users-bounces at lists.opensips.org <mailto:users-bounces at lists.opensips.org>> au nom de Liviu Chircu 
> <liviu at opensips.org <mailto:liviu at opensips.org>>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> *Date : *mercredi 19 juin 2024 à 15:06
> *À : *OpenSIPS Users <users at lists.opensips.org <mailto:users at lists.opensips.org>>
> *Objet : *Re: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.6 
> and 3.2.19 Minor Releases
> 
> Hello,
> 
> A new round of stable minor releases is now out: *3.4.6 *and *3.2.19*. 
> Note that support for *3.2 LTS *release has ended, so make sure to 
> upgrade to *3.4* *LTS* in order to continue receiving fixes.
> 
> Finally, a first stable release candidate for *3.5* branch is now out: 
> *3.5.0-rc1*
> 
> Full changelogs:
> 
> https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog <https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog> 
> <https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog> <https://opensips.org/pub/opensips/3.5.0-rc1/ChangeLog&gt;>
> https://opensips.org/pub/opensips/3.4.6/ChangeLog <https://opensips.org/pub/opensips/3.4.6/ChangeLog> 
> <https://opensips.org/pub/opensips/3.4.6/ChangeLog> <https://opensips.org/pub/opensips/3.4.6/ChangeLog&gt;>
> https://opensips.org/pub/opensips/3.2.19/ChangeLog <https://opensips.org/pub/opensips/3.2.19/ChangeLog> 
> <https://opensips.org/pub/opensips/3.2.19/ChangeLog> <https://opensips.org/pub/opensips/3.2.19/ChangeLog&gt;>
> 
> Please enjoy!
> 
> Liviu Chircu
> 
> www.twitter.com/liviuchircu <http://www.twitter.com/liviuchircu> <http://www.twitter.com/liviuchircu&gt;&nbsp;&nbsp;>|www.opensips-solutions.com <http://www.opensips-solutions.com> <http://www.opensips-solutions.com&gt;>
> 
> On 13.06.2024 17:05, Liviu Chircu wrote:
> 
> Hi all,
> 
> The *3.4.6* and *3.2.19* OpenSIPS minor versions are scheduled for
> release on *Wednesday, June 19th*. Please note that this will mark
> the *end-of-life* for the *3.2* *LTS *version, according to the
> OpenSIPS release policy
> <https://www.opensips.org/Development/Development> <https://www.opensips.org/Development/Development&gt;>.
> 
> Moreover, as the beta testing for release *3.5* is still ongoing, we
> will mark the current progress with a new release candidate: *3.5.0-rc1*
> 
> In preparation for the releases, we impose the usual freeze on any
> significant fixes (as complexity) on the stable branches, in order
> to ensure a safe window for testing in the days ahead.
> 
> Finally, please make sure to ping any outstanding issues on the
> GitHub issue tracker that may have skipped our attention -- /thank
> you/ in advance!
> 
> Happy testing,
> 
> -- 
> 
> Liviu Chircu
> 
> www.twitter.com/liviuchircu <http://www.twitter.com/liviuchircu> <http://www.twitter.com/liviuchircu&gt;&nbsp;&nbsp;>|www.opensips-solutions.com <http://www.opensips-solutions.com> <http://www.opensips-solutions.com&gt;>
> 
> _______________________________________________
> 
> Users mailing list
> 
> Users at lists.opensips.org <mailto:Users at lists.opensips.org> <mailto: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> <http://lists.opensips.org/cgi-bin/mailman/listinfo/users&nbsp;&nbsp;&lt;http://lists.opensips.org/cgi-bin/mailman/listinfo/users&gt;>
> 
> _______________________________________________ Users mailing list 
> Users at lists.opensips.org <mailto:Users at lists.opensips.org> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
> 
> _______________________________________________ Users mailing list 
> Users at lists.opensips.org <mailto:Users at lists.opensips.org> 
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org <mailto:Users at lists.opensips.org>
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users <http://lists.opensips.org/cgi-bin/mailman/listinfo/users>


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






From spanda at 3clogic.com  Fri Jul  5 06:15:20 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 5 Jul 2024 11:45:20 +0530
Subject: [OpenSIPS-Users] I need some help on opensips logging behaviour .
Message-ID: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>

Hi All ,
I want Openisps to have standard error output. In addition to that it , I
want opensips to write the messages on a file as well . Is that possible
anyway ?

log_stderror=yes
log_facility=LOG_LOCAL5

I have this configuration on my config file .  This is not working for me .
What is the configuration on which I will get console output as well as
logging on the log file ?

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From bogdan at opensips.org  Fri Jul  5 06:20:08 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 09:20:08 +0300
Subject: [OpenSIPS-Users] I need some help on opensips logging behaviour
	.
In-Reply-To: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>
References: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>
Message-ID: <825d9dfa-bfff-4fc0-8eb9-08de74b41d32@opensips.org>

Hi,

What opensips version do you have ?

Regards,

Bogdan-Andrei Iancu

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

On 05.07.2024 09:15, Sasmita Panda wrote:
> Hi All ,
> I want Openisps to have standard error output. In addition to that it 
> , I want opensips to write the messages on a file as well . Is that 
> possible anyway ?
>
> log_stderror=yes
> log_facility=LOG_LOCAL5
>
> I have this configuration on my config file .  This is not working for 
> me . What is the configuration on which I will get console output as 
> well as logging on the log file ?
>
> */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 bogdan at opensips.org  Fri Jul  5 06:25:20 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 09:25:20 +0300
Subject: [OpenSIPS-Users] I need help on opensips-trap and opensips-dbg
 package for debugging .
In-Reply-To: <CALf-Q46=9_duNDpxW0YY7e4gHmS2Qgxszyx7zRq2tbO8CnjpVA@mail.gmail.com>
References: <CALf-Q46=9_duNDpxW0YY7e4gHmS2Qgxszyx7zRq2tbO8CnjpVA@mail.gmail.com>
Message-ID: <9ce23d0a-511b-419e-9aaf-ddd34cf67ad4@opensips.org>

Hi Sasmita,

The `trap` cmd in `opensips-cli` depends on gdb being installed - be 
sure you have it in place.

Regards,

Bogdan-Andrei Iancu

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

On 01.07.2024 12:08, Sasmita Panda wrote:
> Hi All ,
>
> I am using the below  linux version and amd base architecture .
>
> [ opensips-codechanged-3.2]# cat /etc/*release
> Amazon Linux release 2023.4.20240401 (Amazon Linux)
> NAME="Amazon Linux"
> VERSION="2023"
> ID="amzn"
> ID_LIKE="fedora"
> VERSION_ID="2023"
> PLATFORM_ID="platform:al2023"
> PRETTY_NAME="Amazon Linux 2023.4.20240401"
> ANSI_COLOR="0;33"
> CPE_NAME="cpe:2.3:o:amazon:amazon_linux:2023"
> HOME_URL="https://aws.amazon.com/linux/amazon-linux-2023/"
> DOCUMENTATION_URL="https://docs.aws.amazon.com/linux/"
> SUPPORT_URL="https://aws.amazon.com/premiumsupport/"
> BUG_REPORT_URL="https://github.com/amazonlinux/amazon-linux-2023"
> VENDOR_NAME="AWS"
> VENDOR_URL="https://aws.amazon.com/"
> SUPPORT_END="2028-03-15"
> Amazon Linux release 2023.4.20240401 (Amazon Linux)
> [ opensips-codechanged-3.2]#
> [ opensips-codechanged-3.2]# uname -r
> 6.1.82-99.168.amzn2023.x86_64
> [ opensips-codechanged-3.2]#
>
> I am getting timer waring while starting opensips which I have posted 
> in the forum earlier as well . But not getting any proper solution for 
> this . Now this is becoming critical for me . I wanted to take core 
> file with backtrace also earlier I got suggestion to take opensips 
> trap command . But opensips-cli says no trap module loaded and I am 
> also not able to install opensips-dbg package on this system .
>
> Is there any proper guideline to install opensips-cli with trap module 
> and opensips-dbg package on the above linux . If not then what is the 
> best suitable version of linux I must use [for opensips where I can 
> install them easily . Please do suggest .
>
>
> Attached the installation doc which I used to follow to install 
> opensips and opensips-cli manually . Please suggest to me what I 
> should do here .
>
>
> */
> /*
> */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 bogdan at opensips.org  Fri Jul  5 06:28:06 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 09:28:06 +0300
Subject: [OpenSIPS-Users] BYE CDR Question
In-Reply-To: <CALLkTp0nWSCo4W1tU1XhyEmPmZp5fEoaB1h4Ytyww56Zx+b7JA@mail.gmail.com>
References: <CALLkTp0nWSCo4W1tU1XhyEmPmZp5fEoaB1h4Ytyww56Zx+b7JA@mail.gmail.com>
Message-ID: <e71c5090-8c15-4e6c-85b4-d8c7d679dfaa@opensips.org>

Hi Alex,

There is something confusing in your report. A CDR (Call Data Record) is 
a per call kind of data. So you cannot have a BYE or INVITE CDR. The CDR 
is of a call. For EVI, do you use the E_ACC_CDR ?

Regards,

Bogdan-Andrei Iancu

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

On 21.06.2024 06:28, Alexander Perkins wrote:
> Hi All. We are using the event interface (evi) to capture CDR data. 
> When testing calls one at a time, we noticed that, if answered, we 
> will get one INVITE CDR, but around four or five BYE CDRs. The four or 
> five BYE CDRs also happens if we don’t answer the call.  Any idea why 
> this happens and how to only get one BYE CDR instead of multiple?
>
> Thank you,
> Alex
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From spanda at 3clogic.com  Fri Jul  5 08:05:41 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 5 Jul 2024 13:35:41 +0530
Subject: [OpenSIPS-Users] I need some help on opensips logging behaviour
	.
In-Reply-To: <825d9dfa-bfff-4fc0-8eb9-08de74b41d32@opensips.org>
References: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>
 <825d9dfa-bfff-4fc0-8eb9-08de74b41d32@opensips.org>
Message-ID: <CALf-Q45cxHDrR6Bs=LaEFWJiC0rzhT_P59z_x92M1zE8ODVVMQ@mail.gmail.com>

version: opensips 3.2.18 (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: ffdb1b473
main.c compiled on  with gcc 12


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


On Fri, Jul 5, 2024 at 11:50 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> Hi,
>
> What opensips version do you have ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 05.07.2024 09:15, Sasmita Panda wrote:
>
> Hi All ,
> I want Openisps to have standard error output. In addition to that it , I
> want opensips to write the messages on a file as well . Is that possible
> anyway ?
>
> log_stderror=yes
> log_facility=LOG_LOCAL5
>
> I have this configuration on my config file .  This is not working for me
> . What is the configuration on which I will get console output as well as
> logging on the log file ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
> _______________________________________________
> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
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From bogdan at opensips.org  Fri Jul  5 08:53:59 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 11:53:59 +0300
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <DB9P251MB0045AB6E0056157BE57BA2C9FCCE2@DB9P251MB0045.EURP251.PROD.OUTLOOK.COM>
References: <DB9P251MB0045AB6E0056157BE57BA2C9FCCE2@DB9P251MB0045.EURP251.PROD.OUTLOOK.COM>
Message-ID: <8e8911ce-bb96-4dd6-8728-cd4085643505@opensips.org>

Hi Srigo,

You the remove in the right way, nothing more you can do about it. The 
problem is how the remove works and how `stir_shaken_verify()` tests for 
the hdr - the two are incompatible.
So, IMHO, we should remove from the `stir_shaken_verify()` function the 
check on the Identity hdr presence . I just pushed this fix on 
3.4/3.5/master versions.

Regards,

Bogdan-Andrei Iancu

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

On 18.06.2024 09:31, Srigo Kanapathipillai wrote:
> Hi,
>
> I'm encountering an issue with removing an Identity header in OpenSIPS 
> 3.4. Here’s the situation:
>
> 1. An incoming call with an Identity header is received.
> 2. I perform a `stir_shaken_verify()` and remove the Identity header 
> in a request route.
> 3. The call is forwarded to an upstream server, but it fails.
> 4. In the `failure_route`, I need to forward the call to a PSTN number.
>
> 5. Before sending the call to the PSTN (in compliance with French 
> STIR/SHAKEN regulations), I need to sign it with my certificate.
>
> However, when I call `stir_shaken_auth()`, I receive an error -2 
> indicating that the Identity header already exists. Despite running 
> `remove_hf(identity)` before calling this function, the header isn't 
> removed, and `$hdr(identity)` still returns the initial value of the 
> Identity header.
>
> What is the best way to remove the existing Identity header and 
> re-sign the call?
>
> Thank you,
> Srigo
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users



From bogdan at opensips.org  Fri Jul  5 08:58:12 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 11:58:12 +0300
Subject: [OpenSIPS-Users] Compiling modules
In-Reply-To: <CAFjCFz=euw0id0Gj4mkQFH8AVB=G+j5n4ywR=XimU-sNdx1JUA@mail.gmail.com>
References: <CAFjCFz=euw0id0Gj4mkQFH8AVB=G+j5n4ywR=XimU-sNdx1JUA@mail.gmail.com>
Message-ID: <34c35746-793d-4180-adb7-4f4592b9bcec@opensips.org>

Hi Callum,

doing `opensips -V` will show the compile flags, so you can do a diff 
between the flags in the official compiling and your compiling.

Regards,

Bogdan-Andrei Iancu

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

On 11.06.2024 18:35, Callum Guy via Users wrote:
> Hi All,
>
> I'm compiling the load_balancer module with some pretty minor changes 
> however the resulting load_balancer.so is ~500k however the standard 
> release is 120k - a size increase of 4x.
>
> My question is simple - why is my version so much bigger? Are there 
> "make" flags that are used for the official releases which I'm failing 
> to include? I build and run on Almalinux 9, is it a simple matter of 
> the libraries used on the build host?
>
> Not a show stopper in any way but I wanted to ask the question in case 
> I'm about to deploy a less performant module etc.
>
> Thanks,
>
> Callum
>
> View and book here 
> <https://www.x-on.co.uk/access-optimisation#webinar-series>
>
>
> *^0333 332 0000  | x-on.co.uk <https://www.x-on.co.uk>  | **^Practice 
> Index Reviews <https://practiceindex.co.uk/gp/x-on> *
>
> *Our new office address: 22 Riduna Park, Melton IP12 1QT.*
>
> X-on is a trading name of X-on Health Ltd a limited company registered 
> in England and Wales.
> Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, 
> Hampshire, England RG25 2AD. Company Registration No. 2578478.
> The information in this e-mail is confidential and for use by the 
> addressee(s) only. If you are not the intended recipient, please 
> notify X-on immediately on +44(0)333 332 0000 and delete the
> message from your computer. If you are not a named addressee you must 
> not use, disclose, disseminate, distribute, copy, print or reply to 
> this email. Views or opinions expressed by an individual
> within this email may not necessarily reflect the views of X-on or its 
> associated companies. Although X-on routinely screens for viruses, 
> addressees should scan this email and any attachments
> for viruses. X-on makes no representation or warranty as to the 
> absence of viruses in this email or any attachments.
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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From osas at voipembedded.com  Fri Jul  5 09:05:44 2024
From: osas at voipembedded.com (Ovidiu Sas)
Date: Fri, 5 Jul 2024 05:05:44 -0400
Subject: [OpenSIPS-Users] Compiling modules
In-Reply-To: <CAFjCFz=euw0id0Gj4mkQFH8AVB=G+j5n4ywR=XimU-sNdx1JUA@mail.gmail.com>
References: <CAFjCFz=euw0id0Gj4mkQFH8AVB=G+j5n4ywR=XimU-sNdx1JUA@mail.gmail.com>
Message-ID: <CAND0Lkt6UkckpUBr9X-ohbrBQ7a=449bxiTzgHRo5rJPqn69ZQ@mail.gmail.com>

Probably you have the debug symbols.
In the packaged version the debug symbols are stripped out.

-ovidiu

On Tue, Jun 11, 2024 at 17:37 Callum Guy via Users <users at lists.opensips.org>
wrote:

> Hi All,
>
> I'm compiling the load_balancer module with some pretty minor changes
> however the resulting load_balancer.so is ~500k however the standard
> release is 120k - a size increase of 4x.
>
> My question is simple - why is my version so much bigger? Are there
> "make" flags that are used for the official releases which I'm failing to
> include? I build and run on Almalinux 9, is it a simple matter of the
> libraries used on the build host?
>
> Not a show stopper in any way but I wanted to ask the question in case I'm
> about to deploy a less performant module etc.
>
> Thanks,
>
> Callum
>
> View and book here
> <https://www.x-on.co.uk/access-optimisation#webinar-series>
>
>
> *0333 332 0000  |  x-on.co.uk <https://www.x-on.co.uk>  | **Practice
> Index Reviews <https://practiceindex.co.uk/gp/x-on>*
>
> *Our new office address: 22 Riduna Park, Melton IP12 1QT.*
>
> X-on is a trading name of X-on Health Ltd a limited company registered in
> England and Wales.
> Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke,
> Hampshire, England RG25 2AD. Company Registration No. 2578478.
> The information in this e-mail is confidential and for use by the
> addressee(s) only. If you are not the intended recipient, please notify
> X-on immediately on +44(0)333 332 0000 and delete the
> message from your computer. If you are not a named addressee you must not
> use, disclose, disseminate, distribute, copy, print or reply to this email. Views
> or opinions expressed by an individual
> within this email may not necessarily reflect the views of X-on or its
> associated companies. Although X-on routinely screens for viruses,
> addressees should scan this email and any attachments
> for viruses. X-on makes no representation or warranty as to the absence of
> viruses in this email or any attachments.
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From bogdan at opensips.org  Fri Jul  5 09:06:48 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 5 Jul 2024 12:06:48 +0300
Subject: [OpenSIPS-Users] I need some help on opensips logging behaviour
	.
In-Reply-To: <CALf-Q45cxHDrR6Bs=LaEFWJiC0rzhT_P59z_x92M1zE8ODVVMQ@mail.gmail.com>
References: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>
 <825d9dfa-bfff-4fc0-8eb9-08de74b41d32@opensips.org>
 <CALf-Q45cxHDrR6Bs=LaEFWJiC0rzhT_P59z_x92M1zE8ODVVMQ@mail.gmail.com>
Message-ID: <89432643-548f-448d-9221-de5f03f807c6@opensips.org>

With this version you can either log to syslog, either to stderr. You 
cannot log (directly from OpenSIPS) to two destinations in the same time.

Regards,

Bogdan-Andrei Iancu

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

On 05.07.2024 11:05, Sasmita Panda wrote:
> version: opensips 3.2.18 (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: ffdb1b473
> main.c compiled on  with gcc 12
> */
> /*
> */
> /*
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
>
>
> On Fri, Jul 5, 2024 at 11:50 AM Bogdan-Andrei Iancu 
> <bogdan at opensips.org> wrote:
>
>     Hi,
>
>     What opensips version do you have ?
>
>     Regards,
>
>     Bogdan-Andrei Iancu
>
>     OpenSIPS Founder and Developer
>        https://www.opensips-solutions.com
>        https://www.siphub.com
>
>     On 05.07.2024 09:15, Sasmita Panda wrote:
>>     Hi All ,
>>     I want Openisps to have standard error output. In addition to
>>     that it , I want opensips to write the messages on a file as well
>>     . Is that possible anyway ?
>>
>>     log_stderror=yes
>>     log_facility=LOG_LOCAL5
>>
>>     I have this configuration on my config file .  This is not
>>     working for me . What is the configuration on which I will get
>>     console output as well as logging on the log file ?
>>
>>     */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 david.villasmil.work at gmail.com  Fri Jul  5 09:37:10 2024
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 5 Jul 2024 11:37:10 +0200
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <AFAE4681-437A-4D2C-834C-21F960945827@free.fr>
References: <AFAE4681-437A-4D2C-834C-21F960945827@free.fr>
Message-ID: <CAFGRPVr9tXEbh1-N_k7DxZgEVKrEoqOE8VMG3imkFLbtoUiH+Q@mail.gmail.com>

this is really getting ridiculous... and they think they can stop robocalls
with this.. they never will.
Regards,

David Villasmil
email: david.villasmil.work at gmail.com



On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent <alain.bieuzent at free.fr>
wrote:

> Hi,
>
>
>
> interesting question, because in future developments of stir/shaken in
> France, for forwarded calls, it is planned that the identity field received
> on the incoming call be forwarded to the outgoing leg but also to add a
> signature (with the local certificate) on the outgoing call (so two
> identity fields).
>
>
>
> Regards
>
>
>
> *De : *Users <users-bounces at lists.opensips.org> au nom de Srigo
> Kanapathipillai <ksrigo at gmail.com>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *mardi 18 juin 2024 à 08:34
> *À : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Objet : *[OpenSIPS-Users] Removing Identity hdr
>
>
>
> Hi,
>
>
>
> I'm encountering an issue with removing an Identity header in OpenSIPS
> 3.4. Here’s the situation:
>
>
>
> 1. An incoming call with an Identity header is received.
>
> 2. I perform a `stir_shaken_verify()` and remove the Identity header in a
> request route.
>
> 3. The call is forwarded to an upstream server, but it fails.
>
> 4. In the `failure_route`, I need to forward the call to a PSTN number.
>
>
>
> 5. Before sending the call to the PSTN (in compliance with French
> STIR/SHAKEN regulations), I need to sign it with my certificate.
>
>
>
> However, when I call `stir_shaken_auth()`, I receive an error -2
> indicating that the Identity header already exists. Despite running
> `remove_hf(identity)` before calling this function, the header isn't
> removed, and `$hdr(identity)` still returns the initial value of the
> Identity header.
>
>
>
> What is the best way to remove the existing Identity header and re-sign
> the call?
>
>
>
> Thank you,
>
> Srigo
>
> _______________________________________________ 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  Fri Jul  5 09:58:42 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 5 Jul 2024 15:28:42 +0530
Subject: [OpenSIPS-Users] I need some help on opensips logging behaviour
	.
In-Reply-To: <89432643-548f-448d-9221-de5f03f807c6@opensips.org>
References: <CALf-Q45hXY=JpDy4m0e3Mz+5iXqbPRcW=R9m3Sk0Jad27k9_+Q@mail.gmail.com>
 <825d9dfa-bfff-4fc0-8eb9-08de74b41d32@opensips.org>
 <CALf-Q45cxHDrR6Bs=LaEFWJiC0rzhT_P59z_x92M1zE8ODVVMQ@mail.gmail.com>
 <89432643-548f-448d-9221-de5f03f807c6@opensips.org>
Message-ID: <CALf-Q44Qsew8_6RP_Y8xGZjaxjpaekZhErsgFVM5oowjQqE-ew@mail.gmail.com>

Ok . Thank you.

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


On Fri, Jul 5, 2024 at 2:36 PM Bogdan-Andrei Iancu <bogdan at opensips.org>
wrote:

> With this version you can either log to syslog, either to stderr. You
> cannot log (directly from OpenSIPS) to two destinations in the same time.
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>   https://www.opensips-solutions.com
>   https://www.siphub.com
>
> On 05.07.2024 11:05, Sasmita Panda wrote:
>
> version: opensips 3.2.18 (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: ffdb1b473
> main.c compiled on  with gcc 12
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Fri, Jul 5, 2024 at 11:50 AM Bogdan-Andrei Iancu <bogdan at opensips.org>
> wrote:
>
>> Hi,
>>
>> What opensips version do you have ?
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>>   https://www.opensips-solutions.com
>>   https://www.siphub.com
>>
>> On 05.07.2024 09:15, Sasmita Panda wrote:
>>
>> Hi All ,
>> I want Openisps to have standard error output. In addition to that it , I
>> want opensips to write the messages on a file as well . Is that possible
>> anyway ?
>>
>> log_stderror=yes
>> log_facility=LOG_LOCAL5
>>
>> I have this configuration on my config file .  This is not working for me
>> . What is the configuration on which I will get console output as well as
>> logging on the log file ?
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>>
>> _______________________________________________
>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>
>
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From alain.bieuzent at free.fr  Fri Jul  5 11:35:49 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Fri, 05 Jul 2024 13:35:49 +0200
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <CAFGRPVr9tXEbh1-N_k7DxZgEVKrEoqOE8VMG3imkFLbtoUiH+Q@mail.gmail.com>
References: <AFAE4681-437A-4D2C-834C-21F960945827@free.fr>
 <CAFGRPVr9tXEbh1-N_k7DxZgEVKrEoqOE8VMG3imkFLbtoUiH+Q@mail.gmail.com>
Message-ID: <AE09D407-A0B9-4C9B-8A1A-0A4C36E0833E@free.fr>

Hi David,

 

The implementation of stir-shaken in France is different from in the US.

The text which requires French operators to implement stir-shaken aims to stop the usurpation of caller-id.

So, what is asked of the operator is to check their customer's caller-id and sign outgoing calls, in the event of fraud, it will then be easy to trace the malicious operator.

For operators receiving unsigned or incorrectly signed traffic, the call must be disconnected.

 

for the case of call forwarding (A -> B then B-> C), there will therefore be two signatures, a first issued by the operator of A and which will be controlled by the operator of B. Then operator B will add his own signature (in addition to that of A), both signatures will be controlled by C

 

Regards

De : Users <users-bounces at lists.opensips.org> au nom de David Villasmil <david.villasmil.work at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : vendredi 5 juillet 2024 à 11:39
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Removing Identity hdr

 

this is really getting ridiculous... and they think they can stop robocalls with this.. they never will.
Regards,

 

David Villasmil

email: david.villasmil.work at gmail.com

 

 

 

On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent <alain.bieuzent at free.fr> wrote:

Hi,

 

interesting question, because in future developments of stir/shaken in France, for forwarded calls, it is planned that the identity field received on the incoming call be forwarded to the outgoing leg but also to add a signature (with the local certificate) on the outgoing call (so two identity fields).

 

Regards

 

De : Users <users-bounces at lists.opensips.org> au nom de Srigo Kanapathipillai <ksrigo at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 18 juin 2024 à 08:34
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : [OpenSIPS-Users] Removing Identity hdr

 

Hi,

 

I'm encountering an issue with removing an Identity header in OpenSIPS 3.4. Here’s the situation:

 

1. An incoming call with an Identity header is received.

2. I perform a `stir_shaken_verify()` and remove the Identity header in a request route.

3. The call is forwarded to an upstream server, but it fails.

4. In the `failure_route`, I need to forward the call to a PSTN number. 

 

5. Before sending the call to the PSTN (in compliance with French STIR/SHAKEN regulations), I need to sign it with my certificate.

 

However, when I call `stir_shaken_auth()`, I receive an error -2 indicating that the Identity header already exists. Despite running `remove_hf(identity)` before calling this function, the header isn't removed, and `$hdr(identity)` still returns the initial value of the Identity header.

 

What is the best way to remove the existing Identity header and re-sign the call?

 

Thank you,

Srigo

_______________________________________________ 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 david.villasmil.work at gmail.com  Fri Jul  5 11:44:29 2024
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 5 Jul 2024 13:44:29 +0200
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <AE09D407-A0B9-4C9B-8A1A-0A4C36E0833E@free.fr>
References: <AFAE4681-437A-4D2C-834C-21F960945827@free.fr>
 <CAFGRPVr9tXEbh1-N_k7DxZgEVKrEoqOE8VMG3imkFLbtoUiH+Q@mail.gmail.com>
 <AE09D407-A0B9-4C9B-8A1A-0A4C36E0833E@free.fr>
Message-ID: <CAFGRPVpncFJakDub63z7ye83uN0d4YkNq9TVztxnXfvXHuR4Fw@mail.gmail.com>

Thanks, yes I know the logic behind it, but nobody is going to reject calls
because they have no caller id signed. That’s money. And this won’t stop
caller id spoofing either. It’s very naive to think so, IMO.

Hopefully it does , though! But telemarketers ingenuity never ceases to
amaze me.

Regards,

David Villasmil
email: david.villasmil.work at gmail.com



On Fri, 5 Jul 2024 at 13:36, Alain Bieuzent <alain.bieuzent at free.fr> wrote:

> Hi David,
>
>
>
> The implementation of stir-shaken in France is different from in the US.
>
> The text which requires French operators to implement stir-shaken aims to
> stop the usurpation of caller-id.
>
> So, what is asked of the operator is to check their customer's caller-id
> and sign outgoing calls, in the event of fraud, it will then be easy to
> trace the malicious operator.
>
> For operators receiving unsigned or incorrectly signed traffic, the call
> must be disconnected.
>
>
>
> for the case of call forwarding (A -> B then B-> C), there will therefore
> be two signatures, a first issued by the operator of A and which will be
> controlled by the operator of B. Then operator B will add his own signature
> (in addition to that of A), both signatures will be controlled by C
>
>
>
> Regards
>
> *De : *Users <users-bounces at lists.opensips.org> au nom de David Villasmil
> <david.villasmil.work at gmail.com>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *vendredi 5 juillet 2024 à 11:39
> *À : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Objet : *Re: [OpenSIPS-Users] Removing Identity hdr
>
>
>
> this is really getting ridiculous... and they think they can stop
> robocalls with this.. they never will.
>
> Regards,
>
>
>
> David Villasmil
>
> email: david.villasmil.work at gmail.com
>
>
>
>
>
>
>
> On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent <alain.bieuzent at free.fr>
> wrote:
>
> Hi,
>
>
>
> interesting question, because in future developments of stir/shaken in
> France, for forwarded calls, it is planned that the identity field received
> on the incoming call be forwarded to the outgoing leg but also to add a
> signature (with the local certificate) on the outgoing call (so two
> identity fields).
>
>
>
> Regards
>
>
>
> *De : *Users <users-bounces at lists.opensips.org> au nom de Srigo
> Kanapathipillai <ksrigo at gmail.com>
> *Répondre à : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Date : *mardi 18 juin 2024 à 08:34
> *À : *OpenSIPS users mailling list <users at lists.opensips.org>
> *Objet : *[OpenSIPS-Users] Removing Identity hdr
>
>
>
> Hi,
>
>
>
> I'm encountering an issue with removing an Identity header in OpenSIPS
> 3.4. Here’s the situation:
>
>
>
> 1. An incoming call with an Identity header is received.
>
> 2. I perform a `stir_shaken_verify()` and remove the Identity header in a
> request route.
>
> 3. The call is forwarded to an upstream server, but it fails.
>
> 4. In the `failure_route`, I need to forward the call to a PSTN number.
>
>
>
> 5. Before sending the call to the PSTN (in compliance with French
> STIR/SHAKEN regulations), I need to sign it with my certificate.
>
>
>
> However, when I call `stir_shaken_auth()`, I receive an error -2
> indicating that the Identity header already exists. Despite running
> `remove_hf(identity)` before calling this function, the header isn't
> removed, and `$hdr(identity)` still returns the initial value of the
> Identity header.
>
>
>
> What is the best way to remove the existing Identity header and re-sign
> the call?
>
>
>
> Thank you,
>
> Srigo
>
> _______________________________________________ Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> _______________________________________________ Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From alain.bieuzent at free.fr  Fri Jul  5 12:08:44 2024
From: alain.bieuzent at free.fr (Alain Bieuzent)
Date: Fri, 05 Jul 2024 14:08:44 +0200
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <CAFGRPVpncFJakDub63z7ye83uN0d4YkNq9TVztxnXfvXHuR4Fw@mail.gmail.com>
References: <AFAE4681-437A-4D2C-834C-21F960945827@free.fr>
 <CAFGRPVr9tXEbh1-N_k7DxZgEVKrEoqOE8VMG3imkFLbtoUiH+Q@mail.gmail.com>
 <AE09D407-A0B9-4C9B-8A1A-0A4C36E0833E@free.fr>
 <CAFGRPVpncFJakDub63z7ye83uN0d4YkNq9TVztxnXfvXHuR4Fw@mail.gmail.com>
Message-ID: <B8256C4C-770F-434E-9100-A71347720BC4@free.fr>

“nobody is going to reject calls because they have no caller id signed”, there is a lot of pressure from the banks, in particular, for calls to be cut. The real cut starts on October 1st, we'll see.

The operator who does not cut calls will find himself outside the law.

 

“That’s money » , yes and no, the rollover on incoming calls is so low, that no operator can live with that.

 

Regards

 

De : Users <users-bounces at lists.opensips.org> au nom de David Villasmil <david.villasmil.work at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : vendredi 5 juillet 2024 à 13:47
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Removing Identity hdr

 

Thanks, yes I know the logic behind it, but nobody is going to reject calls because they have no caller id signed. That’s money. And this won’t stop caller id spoofing either. It’s very naive to think so, IMO.

 

Hopefully it does , though! But telemarketers ingenuity never ceases to amaze me.


Regards,

 

David Villasmil

email: david.villasmil.work at gmail.com

 

 

 

On Fri, 5 Jul 2024 at 13:36, Alain Bieuzent <alain.bieuzent at free.fr> wrote:

Hi David,

 

The implementation of stir-shaken in France is different from in the US.

The text which requires French operators to implement stir-shaken aims to stop the usurpation of caller-id.

So, what is asked of the operator is to check their customer's caller-id and sign outgoing calls, in the event of fraud, it will then be easy to trace the malicious operator.

For operators receiving unsigned or incorrectly signed traffic, the call must be disconnected.

 

for the case of call forwarding (A -> B then B-> C), there will therefore be two signatures, a first issued by the operator of A and which will be controlled by the operator of B. Then operator B will add his own signature (in addition to that of A), both signatures will be controlled by C

 

Regards

De : Users <users-bounces at lists.opensips.org> au nom de David Villasmil <david.villasmil.work at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : vendredi 5 juillet 2024 à 11:39
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : Re: [OpenSIPS-Users] Removing Identity hdr

 

this is really getting ridiculous... and they think they can stop robocalls with this.. they never will.
Regards,

 

David Villasmil

email: david.villasmil.work at gmail.com

 

 

 

On Tue, Jun 18, 2024 at 10:56 AM Alain Bieuzent <alain.bieuzent at free.fr> wrote:

Hi,

 

interesting question, because in future developments of stir/shaken in France, for forwarded calls, it is planned that the identity field received on the incoming call be forwarded to the outgoing leg but also to add a signature (with the local certificate) on the outgoing call (so two identity fields).

 

Regards

 

De : Users <users-bounces at lists.opensips.org> au nom de Srigo Kanapathipillai <ksrigo at gmail.com>
Répondre à : OpenSIPS users mailling list <users at lists.opensips.org>
Date : mardi 18 juin 2024 à 08:34
À : OpenSIPS users mailling list <users at lists.opensips.org>
Objet : [OpenSIPS-Users] Removing Identity hdr

 

Hi,

 

I'm encountering an issue with removing an Identity header in OpenSIPS 3.4. Here’s the situation:

 

1. An incoming call with an Identity header is received.

2. I perform a `stir_shaken_verify()` and remove the Identity header in a request route.

3. The call is forwarded to an upstream server, but it fails.

4. In the `failure_route`, I need to forward the call to a PSTN number. 

 

5. Before sending the call to the PSTN (in compliance with French STIR/SHAKEN regulations), I need to sign it with my certificate.

 

However, when I call `stir_shaken_auth()`, I receive an error -2 indicating that the Identity header already exists. Despite running `remove_hf(identity)` before calling this function, the header isn't removed, and `$hdr(identity)` still returns the initial value of the Identity header.

 

What is the best way to remove the existing Identity header and re-sign the call?

 

Thank you,

Srigo

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

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

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

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

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

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From bogdan at opensips.org  Mon Jul  8 12:12:19 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Mon, 8 Jul 2024 15:12:19 +0300
Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training 2024
Message-ID: <18b700a9-0b1c-4783-a1fa-02a5b416c2fb@opensips.org>

<https://opensips.org/training/OpenSIPS_eBootcamp_2024/>


      14th - 25th October 2024,


          online, worldwide



*Study smarter, not harder!
*

Take advantage of the *OpenSIPS Bootcamp* 
<https://opensips.org/training/OpenSIPS_eBootcamp_2024> and improve your 
OpenSIPS skills - an in-cloud training, a ten days, 4 hours per day (40 
hours) intensive and practical training, covering installation, 
configuration and administration on OpenSIPS.

All the knowledge transferred to the students will be strongly backed up 
by practice sessions where you will get hands-on experience in handling 
OpenSIPS. The training is structured to be offer 50% / 50% between the 
theoretical and practical sessions.

Check Syllabus 
<https://opensips.org/training/OpenSIPS_eBootcamp_2024/#mu-schedule>

*Early Birds open*

The Early Bird 10% discount is available for registrations before /*31st 
of July 2024*/, so do not miss the opportunity. The number of seats is 
limited, so be sure and book a seat now. Keep in mind that a 10% group 
discount is also available - grab your work mate and start learning more 
OpenSIPS together .

Register Now 
<https://opensips.org/training/OpenSIPS_eBootcamp_2024/#mu-pricing>

*Certified training saves time and money*

OpenSIPS mistakes are easily avoided if you get proper training! 
Companies that use OpenSIPS waste time and money when they don't have a 
trained engineer on staff. Searching on Google, waiting on IRC, even the 
latency in mailing list replies takes it's toll over time. Take this 
rare opportunity to train your employees with the project members 
themselves.


Any questions? do not hesitate to contact us <mailto:bootcamp at opensips.org>!

------------------------------------------------------------------------
You received this email as part of your relationship with the OpenSIPS 
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 carsten.bock.private at gmail.com  Tue Jul  9 11:18:08 2024
From: carsten.bock.private at gmail.com (Carsten Bock)
Date: Tue, 9 Jul 2024 13:18:08 +0200
Subject: [OpenSIPS-Users] AAA_diameter usage
Message-ID: <CADw-rRx4cNJL2jasaxbtEoWdaKqpCiKmkLZn8OvX=asacyCFEQ@mail.gmail.com>

Hi,

I've been playing around with the latest aaa_diameter module from OpenSIPS
master. I am currently sending authentications using Diameter (MAR), which
works just fine, and I have also tried to extend this to send a SAR request.

>From my config:
loadmodule "aaa_diameter.so"
modparam("aaa_diameter", "realm", "ims.mnc001.mcc001.3gppnetwork.org")
modparam("aaa_diameter", "peer_identity", "
scscf-1-dev.mnc001.mcc001.3gppnetwork.org")
modparam("aaa_diameter", "fd_log_level", 0)
# modparam("aaa_diameter", "aaa_url",
"diameter:/etc/opensips/freeDiameter.conf;extra-avps-file:/etc/opensips/aka_av_diameter.dictionary")

loadmodule "auth.so"
loadmodule "auth_aka.so"
modparam("auth_aka", "default_av_mgm", "diameter")
modparam("auth_aka", "default_qop", "auth,auth-int")
modparam("auth_aka", "default_algorithm", "AKAv1-MD5")

loadmodule "aka_av_diameter.so"
modparam("aka_av_diameter", "aaa_url",
"diameter:/etc/opensips/freeDiameter.conf;extra-avps-file:/etc/opensips/aka_av_diameter.dictionary"
)
modparam("aka_av_diameter", "realm", "ims.mnc001.mcc001.3gppnetwork.org")

The first thing I've noticed is that the current master crashes if I define
two different "aaa_url"s, e.g., in "aka_av_diameter" for MAR and
"aaa_diameter" for all other Diameter requests. However, if I only define
the "aaa_url" for the "aka_av_diameter" module, OpenSIPS seems fine.

My Diameter Config is limited to only basic stack configuration and a
single peer (the HSS).

For sending a SAR request, I've extended the dictionary accordingly:
ATTRIBUTE Server-Assignment-Type              614 integer  10415
ATTRIBUTE User-Data-Already-Available         624 integer  10415
ATTRIBUTE Cx-User-Data  606 string 10415

(TS 29.229 17.2 mentions "Server-Assignment-Type" and
"User-Data-Already-Available" types should be an Enumeration, however
looking at "app_opensips/avps.c" from the aaa_diameter module indicates
that enums are internally handled as integers, so I used integers instead)

When adding these attributes to the SAR request, OpenSIPS fails to start,
with meaningless errors.

REQUEST 301 Server-Assignment Request
{
Session-Id | REQUIRED | 1
Origin-Host | REQUIRED | 1
Origin-Realm | REQUIRED | 1
Destination-Realm | REQUIRED | 1
Auth-Session-State | REQUIRED | 1
User-Name | REQUIRED | 1
User-Data-Already-Available | REQUIRED | 1
Server-Assignment-Type | REQUIRED | 1
Public-Identity | REQUIRED | 1
Server-Name | REQUIRED | 1
}

The definition itself seems to be fine: If I rename the
"Server-Assignment-Type" to "SAT" and "User-Data-Already-Available" to
"UDA-Available", OpenSIPS starts. However, if I follow the examples (e.g.
module docs for aaa_diameter and here
https://www.opensips.org/Documentation/Tutorials-Diameter-Client-Server), I
fail to send the Diameter-Request:

11:10:14  ERROR  ERROR: Invalid parameter '(((avp) && (((struct
msg_avp_chain *)(avp))->type == MSG_AVP) && (((struct avp
*)(avp))->avp_eyec == (0x11355467))) && pdata)', 22

Am I missing something? Can someone share some example code for sending a
SAR request? Is the documentation missing something?

Thanks,
Carsten
--
Schöne Grüße aus Hamburg, dem Tor zur Welt,
Carsten Bock

T +49 179 2021244 I carsten at bock.info
LinkedIn: https://www.linkedin.com/in/carstenbock/
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From liviu at opensips.org  Tue Jul  9 14:28:24 2024
From: liviu at opensips.org (Liviu Chircu)
Date: Tue, 9 Jul 2024 17:28:24 +0300
Subject: [OpenSIPS-Users] AAA_diameter usage
In-Reply-To: <CADw-rRx4cNJL2jasaxbtEoWdaKqpCiKmkLZn8OvX=asacyCFEQ@mail.gmail.com>
References: <CADw-rRx4cNJL2jasaxbtEoWdaKqpCiKmkLZn8OvX=asacyCFEQ@mail.gmail.com>
Message-ID: <1a525719-f6c1-4388-4faa-283029bea110@opensips.org>

On 09.07.2024 14:18, Carsten Bock wrote:
> The definition itself seems to be fine: If I rename the 
> "Server-Assignment-Type" to "SAT" and "User-Data-Already-Available" to 
> "UDA-Available", OpenSIPS starts. However, if I follow the examples 
> (e.g. module docs for aaa_diameter and here 
> https://www.opensips.org/Documentation/Tutorials-Diameter-Client-Server), 
> I fail to send the Diameter-Request:
>
> 11:10:14  ERROR  ERROR: Invalid parameter '(((avp) && (((struct 
> msg_avp_chain *)(avp))->type == MSG_AVP) && (((struct avp 
> *)(avp))->avp_eyec == (0x11355467))) && pdata)', 22
>
> Am I missing something? Can someone share some example code for 
> sending a SAR request? Is the documentation missing something?

Hello Carsten,

It is great to hear you are having fun with the module!  With regards to 
freeDiameter library errors (which are taken straight up from 
/usr/include/asm-generic/errno-base.h, btw), I've typically seen two 
codes being returned by the library:

#define EINVAL      22  /* Invalid argument */   -- most frequent
#define EEXIST      17  /* File exists */

Now, with *17* (already exists), this is returned if you are trying to 
register a *duplicate* AVP as string name, but with different properties 
(maybe diff type? diff code?  etc.).  As a general rule, the library 
won't complain if we define a perfectly (?) identical AVP twice - it 
will simply move on and return *0 *(success).

In your case, I suspect that the *22* (invalid arg) is still connected 
to AVP duplication somehow, but the library returns *22* from some other 
part of the error-handling code, as a small mistake.  Especially since 
you're saying /it works/ once you fix the name to something else.

To begin the troubleshooting, which dictionaries / fd extensions are you 
loading right now?  Because if you're loading the *dict_dcca_3gpp.fdx* 
module, this one should already contain both of your AVPs (hence 
explaining your errors!), check their definitions here on the 
/libfreeDiameter/ GitHub:

Server-Assignment-Type 
<https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L6999> 
and User-Data-Already-Available 
<https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L7138>

Best regards,

-- 
Liviu Chircu
www.twitter.com/liviuchircu  |www.opensips-solutions.com
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From spanda at 3clogic.com  Wed Jul 10 05:28:29 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 10 Jul 2024 10:58:29 +0530
Subject: [OpenSIPS-Users] I need some info regarding openisps_fifo file .
Message-ID: <CALf-Q45-t40nh33CZmxkJb0Ew7zbceQhyH0-Lhn_dmJp5GQuqQ@mail.gmail.com>

HI ,

I am using opensips 3.2 .  I have created my fifo file on
/tmp/openisps_fifo .
When I started the service the fifo file was created successfully . But
after some days that file was getting deleted automatically . Due to which
I am not able to run the cli commands .

What is the reason behind deletion of the fifo file ? How do I make this
permanent  ? Or there is any other way to run cli commands to get the
location table data . I am using "*single-instance-no-db " this mode for
usrloc . *

*There is no impact on call but I want to monitor  the location table data
frequently to see if the userAgent is registered or not . I was fetching
this with opensips-cli command . But without opensips_fifo file the command
is not running . *

*Please do help . *

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From slackway2me at gmail.com  Wed Jul 10 09:36:37 2024
From: slackway2me at gmail.com (Alexey)
Date: Wed, 10 Jul 2024 14:36:37 +0500
Subject: [OpenSIPS-Users] I need some info regarding openisps_fifo file .
In-Reply-To: <CALf-Q45-t40nh33CZmxkJb0Ew7zbceQhyH0-Lhn_dmJp5GQuqQ@mail.gmail.com>
References: <CALf-Q45-t40nh33CZmxkJb0Ew7zbceQhyH0-Lhn_dmJp5GQuqQ@mail.gmail.com>
Message-ID: <CAPc6LUGKbu9htd8MMQ5EE0afO-Ve_E0vZbkvcBrDAOijaW_DQQ@mail.gmail.com>

Hello,

add to  /usr/lib/tmpfiles.d/tmp.conf this line:

    # Disable auto-remove of /tmp/opensips_fifo
    x /tmp/opensips_fifo

[1] https://alexeyka.zantsev.com/?p=1252

-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From carsten.bock.private at gmail.com  Wed Jul 10 10:22:41 2024
From: carsten.bock.private at gmail.com (Carsten Bock)
Date: Wed, 10 Jul 2024 12:22:41 +0200
Subject: [OpenSIPS-Users] AAA_diameter usage
In-Reply-To: <1a525719-f6c1-4388-4faa-283029bea110@opensips.org>
References: <CADw-rRx4cNJL2jasaxbtEoWdaKqpCiKmkLZn8OvX=asacyCFEQ@mail.gmail.com>
 <1a525719-f6c1-4388-4faa-283029bea110@opensips.org>
Message-ID: <CADw-rRx0_G_iMYcpV7iC8wjbagaSM3w5a13LJ_DP=kzV_kH-dw@mail.gmail.com>

Hi Liviu,

Thanks for getting back to me and for pointing me in some directions.

So far, I did not loaded any extensions. Upon checking with
dict_dcca_3gpp.fdx I've noticed that the extensions (and the lib) are
initialized in the worker process, while the "extra-avps-file" is loaded
from mod_init (so before).

If I move the loading of the extra-avps-file to the worker process (so
after loading extensions), the aka_av_diameter module complains about not
finding the MAR request. After fixing this, there are apparently some
conflicts in the custom-avp's registered by OpenSIPS and the AVPs defined
in the extension dict_dcca_3gpp.fdx or it's dependencies....

I will get there ;-)

Anyway, pointing me in the right direction was already very helpful!

Thanks,
Carsten
--
Schöne Grüße aus Hamburg, dem Tor zur Welt,
Carsten Bock

Baron-Voght-Str. 128a I 22607 Hamburg I Germany
T +49 179 2021244 I carsten at bock.info
LinkedIn: https://www.linkedin.com/in/carstenbock/


Am Di., 9. Juli 2024 um 16:28 Uhr schrieb Liviu Chircu <liviu at opensips.org>:

> On 09.07.2024 14:18, Carsten Bock wrote:
>
> The definition itself seems to be fine: If I rename the
> "Server-Assignment-Type" to "SAT" and "User-Data-Already-Available" to
> "UDA-Available", OpenSIPS starts. However, if I follow the examples (e.g.
> module docs for aaa_diameter and here
> https://www.opensips.org/Documentation/Tutorials-Diameter-Client-Server),
> I fail to send the Diameter-Request:
>
> 11:10:14  ERROR  ERROR: Invalid parameter '(((avp) && (((struct
> msg_avp_chain *)(avp))->type == MSG_AVP) && (((struct avp
> *)(avp))->avp_eyec == (0x11355467))) && pdata)', 22
>
> Am I missing something? Can someone share some example code for sending a
> SAR request? Is the documentation missing something?
>
> Hello Carsten,
>
> It is great to hear you are having fun with the module!  With regards to
> freeDiameter library errors (which are taken straight up from
> /usr/include/asm-generic/errno-base.h, btw), I've typically seen two codes
> being returned by the library:
>
> #define EINVAL      22  /* Invalid argument */   -- most frequent
> #define EEXIST      17  /* File exists */
>
> Now, with *17* (already exists), this is returned if you are trying to
> register a *duplicate* AVP as string name, but with different properties
> (maybe diff type?  diff code?  etc.).  As a general rule, the library won't
> complain if we define a perfectly (?) identical AVP twice - it will simply
> move on and return *0 *(success).
>
> In your case, I suspect that the *22* (invalid arg) is still connected to
> AVP duplication somehow, but the library returns *22* from some other
> part of the error-handling code, as a small mistake.  Especially since
> you're saying *it works* once you fix the name to something else.
>
> To begin the troubleshooting, which dictionaries / fd extensions are you
> loading right now?  Because if you're loading the *dict_dcca_3gpp.fdx*
> module, this one should already contain both of your AVPs (hence explaining
> your errors!), check their definitions here on the *libfreeDiameter*
> GitHub:
>
> Server-Assignment-Type
> <https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L6999>
> and User-Data-Already-Available
> <https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L7138>
>
> Best regards,
>
> --
> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>
>
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From spanda at 3clogic.com  Thu Jul 11 05:25:39 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Thu, 11 Jul 2024 10:55:39 +0530
Subject: [OpenSIPS-Users] I need some info regarding openisps_fifo file .
In-Reply-To: <CAPc6LUGKbu9htd8MMQ5EE0afO-Ve_E0vZbkvcBrDAOijaW_DQQ@mail.gmail.com>
References: <CALf-Q45-t40nh33CZmxkJb0Ew7zbceQhyH0-Lhn_dmJp5GQuqQ@mail.gmail.com>
 <CAPc6LUGKbu9htd8MMQ5EE0afO-Ve_E0vZbkvcBrDAOijaW_DQQ@mail.gmail.com>
Message-ID: <CALf-Q461GcMTz24Ds3UL6nmAQsye7sKj5GZC5JwhKUhEkiF3BA@mail.gmail.com>

Thanks for the reploy .

For now I have changed the fifo file path in the opensips config to
*"/opt/opensips_fifo
"* and also configured the opensips-cli default fifofile path to
*/opt/opensips_fifo* .

Will that create any problem further ?





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


On Wed, Jul 10, 2024 at 3:09 PM Alexey <slackway2me at gmail.com> wrote:

> Hello,
>
> add to  /usr/lib/tmpfiles.d/tmp.conf this line:
>
>     # Disable auto-remove of /tmp/opensips_fifo
>     x /tmp/opensips_fifo
>
> [1] https://alexeyka.zantsev.com/?p=1252
>
> --
> best regards, 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 bullehs at gmail.com  Thu Jul 11 13:54:47 2024
From: bullehs at gmail.com (HS)
Date: Thu, 11 Jul 2024 18:54:47 +0500
Subject: [OpenSIPS-Users] No Redirection to VM on Busy/Decline
Message-ID: <CA+PgBobwGy=JRxKRknvPM7CKdCbwFCqoYV7BURok487iC+jpEg@mail.gmail.com>

Hi all,

Need some collective wisdom. I am using Opensips 3.0 and have the following
snippet for failure_route:

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


        # redirect the failed to a different VM system
        if (t_check_status("487|408|486|480|603")){
                 rewritehostport("VM.IP:5090");
                t_relay();
                exit;

                # do not set the missed call flag again
        }
}

The call is redirected successfully to the VM if the user is not online.
However, if the user is busy/declines/times out - the call is not
redirected. I get the following error. I have searched quite a few answers,
but don't seem to cover what I seek. Thanks for the help in advance.

Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:tls_conn_init: no TLS client domain found
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new:
failed to do proto 3 specific init for conn 0x7f6f38860760
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
ERROR:proto_tls:proto_tls_send: connect failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() to
VM.IP:5090 for proto tls/3 failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack:
sending request failed
Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay:
t_forward_nonack failed
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From carsten.bock.private at gmail.com  Thu Jul 11 14:49:19 2024
From: carsten.bock.private at gmail.com (Carsten Bock)
Date: Thu, 11 Jul 2024 16:49:19 +0200
Subject: [OpenSIPS-Users] AAA_diameter usage
In-Reply-To: <CADw-rRx0_G_iMYcpV7iC8wjbagaSM3w5a13LJ_DP=kzV_kH-dw@mail.gmail.com>
References: <CADw-rRx4cNJL2jasaxbtEoWdaKqpCiKmkLZn8OvX=asacyCFEQ@mail.gmail.com>
 <1a525719-f6c1-4388-4faa-283029bea110@opensips.org>
 <CADw-rRx0_G_iMYcpV7iC8wjbagaSM3w5a13LJ_DP=kzV_kH-dw@mail.gmail.com>
Message-ID: <CADw-rRyHopNUxnokcg+yyisukFLCYwyLtvbjFrHBm4Qw6B7+UA@mail.gmail.com>

Hi Liviu,

Quick update from my side:
Your initial pointer was correct - the issue concerned the AVP definition.

Extending the dictionary using extensions (like dict_dcca_3gpp) seems
complicated due to the forking and library initialization. I have
temporarily solved this by manually including the dictionary definitions
from dict_dcca_3gpp and its dependencies in the aaa_diameter Module.
Including the dictionaries from the freeDiameter project revealed some
conflicts with the existing AVP definitions from the module, so I removed
those definitions for now.

Duplicating code is not ideal, so this is a temporary solution for me.

I've noticed that the "aka_av_diameter" Module requires some AVPs that
conflict with the standard definitions from the freeDiameter project - I've
noticed that you named some AVPs "3GPP-*". I have replaced these with the
standard definitions from the freeDiameter project and will do some further
testing.

Also, we should improve the way the aaa_diameter module handles situations
when no "Session-Id" is found in the crafted Diameter message. I
accidentally missed adding a "Session-Id"-AVP to my request, and the error
message from OpenSIPS was not very helpful.

I will do some further testing. However, I wanted to highlight that I can
finally send a server-assignment request (SAR) to our HSS!

Thanks again for your hard work, especially for useful feedback.

Thanks,
Carsten



--
Schöne Grüße aus Hamburg, dem Tor zur Welt,
Carsten Bock

Baron-Voght-Str. 128a I 22607 Hamburg I Germany
T +49 179 2021244 I carsten at bock.info
LinkedIn: https://www.linkedin.com/in/carstenbock/


Am Mi., 10. Juli 2024 um 12:22 Uhr schrieb Carsten Bock <
carsten.bock.private at gmail.com>:

> Hi Liviu,
>
> Thanks for getting back to me and for pointing me in some directions.
>
> So far, I did not loaded any extensions. Upon checking with
> dict_dcca_3gpp.fdx I've noticed that the extensions (and the lib) are
> initialized in the worker process, while the "extra-avps-file" is loaded
> from mod_init (so before).
>
> If I move the loading of the extra-avps-file to the worker process (so
> after loading extensions), the aka_av_diameter module complains about not
> finding the MAR request. After fixing this, there are apparently some
> conflicts in the custom-avp's registered by OpenSIPS and the AVPs defined
> in the extension dict_dcca_3gpp.fdx or it's dependencies....
>
> I will get there ;-)
>
> Anyway, pointing me in the right direction was already very helpful!
>
> Thanks,
> Carsten
> --
> Schöne Grüße aus Hamburg, dem Tor zur Welt,
> Carsten Bock
>
> Baron-Voght-Str. 128a I 22607 Hamburg I Germany
> T +49 179 2021244 I carsten at bock.info
> LinkedIn: https://www.linkedin.com/in/carstenbock/
>
>
> Am Di., 9. Juli 2024 um 16:28 Uhr schrieb Liviu Chircu <liviu at opensips.org
> >:
>
>> On 09.07.2024 14:18, Carsten Bock wrote:
>>
>> The definition itself seems to be fine: If I rename the
>> "Server-Assignment-Type" to "SAT" and "User-Data-Already-Available" to
>> "UDA-Available", OpenSIPS starts. However, if I follow the examples (e.g.
>> module docs for aaa_diameter and here
>> https://www.opensips.org/Documentation/Tutorials-Diameter-Client-Server),
>> I fail to send the Diameter-Request:
>>
>> 11:10:14  ERROR  ERROR: Invalid parameter '(((avp) && (((struct
>> msg_avp_chain *)(avp))->type == MSG_AVP) && (((struct avp
>> *)(avp))->avp_eyec == (0x11355467))) && pdata)', 22
>>
>> Am I missing something? Can someone share some example code for sending a
>> SAR request? Is the documentation missing something?
>>
>> Hello Carsten,
>>
>> It is great to hear you are having fun with the module!  With regards to
>> freeDiameter library errors (which are taken straight up from
>> /usr/include/asm-generic/errno-base.h, btw), I've typically seen two codes
>> being returned by the library:
>>
>> #define EINVAL      22  /* Invalid argument */   -- most frequent
>> #define EEXIST      17  /* File exists */
>>
>> Now, with *17* (already exists), this is returned if you are trying to
>> register a *duplicate* AVP as string name, but with different properties
>> (maybe diff type?  diff code?  etc.).  As a general rule, the library won't
>> complain if we define a perfectly (?) identical AVP twice - it will simply
>> move on and return *0 *(success).
>>
>> In your case, I suspect that the *22* (invalid arg) is still connected
>> to AVP duplication somehow, but the library returns *22* from some other
>> part of the error-handling code, as a small mistake.  Especially since
>> you're saying *it works* once you fix the name to something else.
>>
>> To begin the troubleshooting, which dictionaries / fd extensions are you
>> loading right now?  Because if you're loading the *dict_dcca_3gpp.fdx*
>> module, this one should already contain both of your AVPs (hence explaining
>> your errors!), check their definitions here on the *libfreeDiameter*
>> GitHub:
>>
>> Server-Assignment-Type
>> <https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L6999>
>> and User-Data-Already-Available
>> <https://github.com/freeDiameter/freeDiameter/blob/master/extensions/dict_dcca_3gpp/add_avps.c#L7138>
>>
>> Best regards,
>>
>> --
>> Liviu Chircuwww.twitter.com/liviuchircu | www.opensips-solutions.com
>>
>>
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From venefax at gmail.com  Fri Jul 12 05:44:51 2024
From: venefax at gmail.com (Saint Michael)
Date: Fri, 12 Jul 2024 01:44:51 -0400
Subject: [OpenSIPS-Users] This may be a bug
Message-ID: <CAC9cSOCy2LVm=3wqsaPj0g7=Z+Yh6DzpYtxvZcVEiFRqKEV5Jg@mail.gmail.com>

(xlog) NOTICE:ru sip:19206661392 at 1.1.1.1;transport=UDP rU 19206661392
DST= 1.1.1.1 rd= 1.1.1.1
CRITICAL:core:mk_proxy: could not resolve hostname: ";transport=UDP"
ERROR:tm:uri2proxy: bad host name in URI <sip:;transport=UDP:5060>
ERROR:tm:t_forward_nonack: failure to add branches
ERROR:tm:w_t_relay: t_forward_nonack failed
My box is multihomed, and I am using rttpproxy. Since I added a new IP to my box
socket=udp:*:5060 use_workers 80

then I get these errors. The main problem is that my CDR is useless
because if the call fails, the field dst_ip comes as ";transport=UDP"
when it should have been "1.1.1.1"

How do I get around this?


From venefax at gmail.com  Fri Jul 12 05:56:31 2024
From: venefax at gmail.com (Saint Michael)
Date: Fri, 12 Jul 2024 01:56:31 -0400
Subject: [OpenSIPS-Users] Maybe this is a bug version 3.4 latest
Message-ID: <CAC9cSOBf75pS2NLxipk4drgwHmQus8c+GM9s5QBFwJJ3Tz68HQ@mail.gmail.com>

(xlog) NOTICE:ru sip:19206661392 at 1.1.1.1;transport=UDP rU 19206661392
DST= 1.1.1.1 rd= 1.1.1.1
CRITICAL:core:mk_proxy: could not resolve hostname: ";transport=UDP"
ERROR:tm:uri2proxy: bad host name in URI <sip:;transport=UDP:5060>
ERROR:tm:t_forward_nonack: failure to add branches
ERROR:tm:w_t_relay: t_forward_nonack failed
My box is multihomed, and I am using rttpproxy.
this: socket=udp:*:5060 use_workers 80
or this: socket=udp:1.1.1.1:5060 use_workers 80
make no difference
also to use rttpproxy or not makes no difference
The issue seems to be caused by ru
sip:19206661392 at 1.1.1.1;transport=UDP having now the ;transport=UDP,
but I only use RTP, so the tansport= part is irrelevant

The main problem is that my CDR is useless
because if the call fails, the field dst_ip comes as ";transport=UDP"
when it should have been "1.1.1.1"


From venefax at gmail.com  Fri Jul 12 06:20:10 2024
From: venefax at gmail.com (Saint Michael)
Date: Fri, 12 Jul 2024 02:20:10 -0400
Subject: [OpenSIPS-Users] INVITE brings in extra transport=UDP parameter
Message-ID: <CAC9cSOBWO2-ro463-F0L2ZyEhbU=AOmLAzrrzE_uJ2xaUzvAoA@mail.gmail.com>

INVITE sip:19206661392 at 38.95.11.250;transport=UDP
when this happens, opensips has a contaminated ru variable
and since it's read-only, I cannot fix it in code
I tried
if ($ru =~ ";transport=UDP") {
         $ru = $(ru{s.select,0,-13});
but it makes no difference, ru does not change
How do I get around this?


From spanda at 3clogic.com  Fri Jul 12 08:32:50 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 12 Jul 2024 14:02:50 +0530
Subject: [OpenSIPS-Users] I need some help while build opensips from debian
	package .
Message-ID: <CALf-Q44spoB_WHnseQeOcYPibtcEidVzoyPPXYSfDf1BK1J1UQ@mail.gmail.com>

Hi All ,

Below is the server I am using .
ID="ec2"
VERSION="20231013-1532"
PRETTY_NAME="Debian GNU/Linux 12 (bookworm)"
NAME="Debian GNU/Linux"
VERSION_ID="12"
VERSION="12 (bookworm)"
VERSION_CODENAME=bookworm
ID=debian
HOME_URL="https://www.debian.org/"
SUPPORT_URL="https://www.debian.org/support"
BUG_REPORT_URL="https://bugs.debian.org/"


I have installed opensips from the package .

    1  curl https://apt.opensips.org/opensips-org.gpg -o
/usr/share/keyrings/opensips-org.gpg

    2  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
https://apt.opensips.org bookworm 3.2-releases"
>/etc/apt/sources.list.d/opensips.list

    3  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
https://apt.opensips.org bookworm cli-nightly"
>/etc/apt/sources.list.d/opensips-cli.list

   apt update

   apt install opensips-*

Now I wanted to run a config with the httpd module but my service is not
getting started and giving the below error .  httpd.so file was also
present in place . .

*ERROR:httpd:httpd_proc: unable to start http daemon*

What should I do to resolve this ?

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From jeremy at ardley.org  Fri Jul 12 08:56:04 2024
From: jeremy at ardley.org (jeremy ardley)
Date: Fri, 12 Jul 2024 16:56:04 +0800
Subject: [OpenSIPS-Users] I need some help while build opensips from
 debian package .
In-Reply-To: <CALf-Q44spoB_WHnseQeOcYPibtcEidVzoyPPXYSfDf1BK1J1UQ@mail.gmail.com>
References: <CALf-Q44spoB_WHnseQeOcYPibtcEidVzoyPPXYSfDf1BK1J1UQ@mail.gmail.com>
Message-ID: <3f319a65-c855-4336-8aad-4070195fea7a@ardley.org>


On 12/7/24 16:32, Sasmita Panda wrote:
>
> Hi All ,
> Below is the server I am using .
> ID="ec2"
> VERSION="20231013-1532"
> PRETTY_NAME="Debian GNU/Linux 12 (bookworm)"
> NAME="Debian GNU/Linux"
> VERSION_ID="12"
> VERSION="12 (bookworm)"
> VERSION_CODENAME=bookworm
> ID=debian
> HOME_URL="https://www.debian.org/"
> SUPPORT_URL="https://www.debian.org/support"
> BUG_REPORT_URL="https://bugs.debian.org/"
>
>
> I have installed opensips from the package .
>
>     1 curl https://apt.opensips.org/opensips-org.gpg -o 
> /usr/share/keyrings/opensips-org.gpg
>
>     2 echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg] 
> https://apt.opensips.org bookworm 3.2-releases" 
> >/etc/apt/sources.list.d/opensips.list
>
>     3 echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg] 
> https://apt.opensips.org bookworm cli-nightly" 
> >/etc/apt/sources.list.d/opensips-cli.list
>
>
>    apt update
>
>    apt install opensips-*
>
>
> Now I wanted to run a config with the httpd module but my service is 
> not getting started and giving the below error . httpd.so file was 
> also present in place . .
>
> *ERROR:httpd:httpd_proc: unable to start http daemon*
>
> What should I do to resolve this ?
>
> */Thanks & Regards/*
> /Sasmita Panda/
> /Senior Network Testing and Software Engineer/
> /3CLogic , ph:07827611765/
> -- 
Have you checked no other services are running on port 80 and/or 443 ?

What does journalctl / syslog report?
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From spanda at 3clogic.com  Fri Jul 12 13:43:28 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 12 Jul 2024 19:13:28 +0530
Subject: [OpenSIPS-Users] I need some help while build opensips from
 debian package .
In-Reply-To: <3f319a65-c855-4336-8aad-4070195fea7a@ardley.org>
References: <CALf-Q44spoB_WHnseQeOcYPibtcEidVzoyPPXYSfDf1BK1J1UQ@mail.gmail.com>
 <3f319a65-c855-4336-8aad-4070195fea7a@ardley.org>
Message-ID: <CALf-Q47fu2FksrMPX+L3Y09T3vx5W0vrs_RqL0JToW3VN3NZUw@mail.gmail.com>

Thank you so much . I was given 8000 port for httpd module . Somehow that
was getting used for something else .
After changing that to 8001 the service started successfully without any
error .

Thanks again .


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


On Fri, Jul 12, 2024 at 2:29 PM jeremy ardley via Users <
users at lists.opensips.org> wrote:

>
> On 12/7/24 16:32, Sasmita Panda wrote:
>
>
> Hi All ,
>
> Below is the server I am using .
> ID="ec2"
> VERSION="20231013-1532"
> PRETTY_NAME="Debian GNU/Linux 12 (bookworm)"
> NAME="Debian GNU/Linux"
> VERSION_ID="12"
> VERSION="12 (bookworm)"
> VERSION_CODENAME=bookworm
> ID=debian
> HOME_URL="https://www.debian.org/"
> SUPPORT_URL="https://www.debian.org/support"
> BUG_REPORT_URL="https://bugs.debian.org/"
>
>
> I have installed opensips from the package .
>
>     1  curl https://apt.opensips.org/opensips-org.gpg -o
> /usr/share/keyrings/opensips-org.gpg
>
>     2  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
> https://apt.opensips.org bookworm 3.2-releases"
> >/etc/apt/sources.list.d/opensips.list
>
>     3  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
> https://apt.opensips.org bookworm cli-nightly"
> >/etc/apt/sources.list.d/opensips-cli.list
>
>    apt update
>
>    apt install opensips-*
>
> Now I wanted to run a config with the httpd module but my service is not
> getting started and giving the below error .  httpd.so file was also
> present in place . .
>
> *ERROR:httpd:httpd_proc: unable to start http daemon*
>
> What should I do to resolve this ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
> --
>
>
> Have you checked no other services are running on port 80 and/or 443 ?
>
> What does journalctl / syslog report?
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From brett at nemeroff.com  Fri Jul 12 14:18:37 2024
From: brett at nemeroff.com (Brett Nemeroff)
Date: Fri, 12 Jul 2024 09:18:37 -0500
Subject: [OpenSIPS-Users] I need some help while build opensips from
 debian package .
In-Reply-To: <CALf-Q47fu2FksrMPX+L3Y09T3vx5W0vrs_RqL0JToW3VN3NZUw@mail.gmail.com>
References: <CALf-Q44spoB_WHnseQeOcYPibtcEidVzoyPPXYSfDf1BK1J1UQ@mail.gmail.com>
 <3f319a65-c855-4336-8aad-4070195fea7a@ardley.org>
 <CALf-Q47fu2FksrMPX+L3Y09T3vx5W0vrs_RqL0JToW3VN3NZUw@mail.gmail.com>
Message-ID: <CAPwC5wwGO3ivRzKKzqM7KXCB+aNDmoCdmEjusRewiMEmCAXpRg@mail.gmail.com>

Sasmita,
on debian based systems doing something like `netstat -nlp` is really
useful for knowing what processes are using what ports.

You can also use lsof, but I think the netstat trick is easier.
-Brett


On Fri, Jul 12, 2024 at 8:45 AM Sasmita Panda <spanda at 3clogic.com> wrote:

> Thank you so much . I was given 8000 port for httpd module . Somehow that
> was getting used for something else .
> After changing that to 8001 the service started successfully without any
> error .
>
> Thanks again .
>
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Fri, Jul 12, 2024 at 2:29 PM jeremy ardley via Users <
> users at lists.opensips.org> wrote:
>
>>
>> On 12/7/24 16:32, Sasmita Panda wrote:
>>
>>
>> Hi All ,
>>
>> Below is the server I am using .
>> ID="ec2"
>> VERSION="20231013-1532"
>> PRETTY_NAME="Debian GNU/Linux 12 (bookworm)"
>> NAME="Debian GNU/Linux"
>> VERSION_ID="12"
>> VERSION="12 (bookworm)"
>> VERSION_CODENAME=bookworm
>> ID=debian
>> HOME_URL="https://www.debian.org/"
>> SUPPORT_URL="https://www.debian.org/support"
>> BUG_REPORT_URL="https://bugs.debian.org/"
>>
>>
>> I have installed opensips from the package .
>>
>>     1  curl https://apt.opensips.org/opensips-org.gpg -o
>> /usr/share/keyrings/opensips-org.gpg
>>
>>     2  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
>> https://apt.opensips.org bookworm 3.2-releases"
>> >/etc/apt/sources.list.d/opensips.list
>>
>>     3  echo "deb [signed-by=/usr/share/keyrings/opensips-org.gpg]
>> https://apt.opensips.org bookworm cli-nightly"
>> >/etc/apt/sources.list.d/opensips-cli.list
>>
>>    apt update
>>
>>    apt install opensips-*
>>
>> Now I wanted to run a config with the httpd module but my service is not
>> getting started and giving the below error .  httpd.so file was also
>> present in place . .
>>
>> *ERROR:httpd:httpd_proc: unable to start http daemon*
>>
>> What should I do to resolve this ?
>>
>> *Thanks & Regards*
>> *Sasmita Panda*
>> *Senior Network Testing and Software Engineer*
>> *3CLogic , ph:07827611765*
>> --
>>
>>
>> Have you checked no other services are running on port 80 and/or 443 ?
>>
>> What does journalctl / syslog report?
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From mickael at winlux.fr  Wed Jul 17 09:30:24 2024
From: mickael at winlux.fr (Mickael Hubert)
Date: Wed, 17 Jul 2024 11:30:24 +0200
Subject: [OpenSIPS-Users] CANCEL cross 200OK
Message-ID: <CA+fRVSVNY9d=yPr=JT57pgf3VkAwWJWHWJgfW8gDMzgVRZB9XQ@mail.gmail.com>

Hi all,
I have an issue in my OpenSIPS proxy (version: opensips 3.3.4
(x86_64/linux)).
Proxy receives 200OK from UAS, but in the same time, receives CANCEL from
UAC.
Ex:
......
UAS --> 200OK (SDP) --> proxy
                        proxy <--  CANCEL <-- UAC
                        proxy --> 200 CANCELING --> UAC
                        proxy --> 200OK  (SDP) --> UAC
                        proxy <-- ACK <-- UAC
UAS <-- ACK <--         proxy
.....

I want to find a solution that proxy sends 487 to UAC, and BYE to UAS.
How can I do that please ? There is a function or I have to code all this
scenario ?

thanks in advance
Have a good day
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From bogdan at opensips.org  Fri Jul 19 09:41:14 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Fri, 19 Jul 2024 12:41:14 +0300
Subject: [OpenSIPS-Users] [OpenSIPS-Devel] OpenSIPS 3.5.0 major release,
	beta version
In-Reply-To: <ca64308c-a97d-4421-8476-4e0f8ab5aa9b@opensips.org>
References: <ca64308c-a97d-4421-8476-4e0f8ab5aa9b@opensips.org>
Message-ID: <3cbbeb9d-8e76-4370-9699-b358a1f4e9f7@opensips.org>

Heads up, the 3.5.0 stable release is planned for 25th of July 2024.

Do you still have any important issues to report? We are here to fix 
them :).

Best regards,

Bogdan-Andrei Iancu

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

On 09.05.2024 19:45, Bogdan-Andrei Iancu wrote:
> Hello there !!
>
> It is that time of the year to do our iteration - one more year, one 
> more evolution step, one more OpenSIPS major release.
>
> So, we are all happy to announce the beta release of *OpenSIPS 3.5.0 
> major version* - and this 3.5 version is all about IMS, about _AKA 
> authentication_ support, about the _DIAMETER and HTTP/2 IMS 
> interfaces_, about _IPSEC support_ and more. Besides IMS, the 3.5 
> comes with _Launch Darkly_ integration, with _Message Queue_ support, 
> with advanced _SQL operations_ and many more.
>
> But here is the shortest possible description 
> <https://www.opensips.org/About/Version-Overview-3-5-0> of this 
> release; and be aware that it's actually not so short as nothing is 
> short about 3.5 and IMS !
>
> Please keep in mind that 3.5.0 is still a beta release, targeting mid 
> July to become fully stable. So, we still have some testing ahead of us :)
>
> Many thanks to our awesome community for contributing with ideas, 
> code, patches, tests and reports!
>
> Looking for downloading it? See the tarball 
> <http://opensips.org/pub/opensips/3.5.0> or the GIT repo 
> <https://github.com/OpenSIPS/opensips/tree/3.5>.
>
> Enjoy it,
> -- 
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
>    https://www.opensips-solutions.com
>    https://www.siphub.com
>
> _______________________________________________
> Devel mailing list
> Devel at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
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From razvan at opensips.org  Mon Jul 22 06:22:48 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Mon, 22 Jul 2024 09:22:48 +0300
Subject: [OpenSIPS-Users] No Redirection to VM on Busy/Decline
In-Reply-To: <CA+PgBobwGy=JRxKRknvPM7CKdCbwFCqoYV7BURok487iC+jpEg@mail.gmail.com>
References: <CA+PgBobwGy=JRxKRknvPM7CKdCbwFCqoYV7BURok487iC+jpEg@mail.gmail.com>
Message-ID: <7ba4ae48-36cc-408b-b067-f145f219363e@opensips.org>

Your VM is probably reachable over UDP, not TLS - you need to make sure 
that your $ru has a UDP transport, rather than what is leftover from the 
client's uri.

Best regards,

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

On 7/11/24 4:54 PM, HS wrote:
> Hi all,
> 
> Need some collective wisdom. I am using Opensips 3.0 and have the following
> snippet for failure_route:
> 
> failure_route[missed_call] {
>          if (t_was_cancelled()) {
>          rtpengine_delete();
>          exit;
>          }
> 
> 
>          # redirect the failed to a different VM system
>          if (t_check_status("487|408|486|480|603")){
>                   rewritehostport("VM.IP:5090");
>                  t_relay();
>                  exit;
> 
>                  # do not set the missed call flag again
>          }
> }
> 
> The call is redirected successfully to the VM if the user is not online.
> However, if the user is busy/declines/times out - the call is not
> redirected. I get the following error. I have searched quite a few answers,
> but don't seem to cover what I seek. Thanks for the help in advance.
> 
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_conn_init: no TLS client domain found
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new:
> failed to do proto 3 specific init for conn 0x7f6f38860760
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:proto_tls_send: connect failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() to
> VM.IP:5090 for proto tls/3 failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack:
> sending request failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay:
> t_forward_nonack failed
> 
> 
> Hi all,
> 
> Need some collective wisdom. I am using Opensips 3.0 and have the 
> following snippet for failure_route:
> 
> failure_route[missed_call] {
>          if (t_was_cancelled()) {
>          rtpengine_delete();
>          exit;
>          }
> 
> 
>          # redirect the failed to a different VM system
>          if (t_check_status("487|408|486|480|603")){
>                   rewritehostport("VM.IP:5090");
>                  t_relay();
>                  exit;
> 
>                  # do not set the missed call flag again
>          }
> }
> 
> The call is redirected successfully to the VM if the user is not online. 
> However, if the user is busy/declines/times out - the call is not 
> redirected. I get the following error. I have searched quite a few 
> answers, but don't seem to cover what I seek. Thanks for the help in 
> advance.
> 
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: 
> ERROR:proto_tls:tls_conn_init: no TLS client domain found
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new: 
> failed to do proto 3 specific init for conn 0x7f6f38860760
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: 
> ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: 
> ERROR:proto_tls:proto_tls_send: connect failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() 
> to VM.IP:5090 for proto tls/3 failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack: 
> sending request failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay: 
> t_forward_nonack failed
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From bullehs at gmail.com  Mon Jul 22 10:55:35 2024
From: bullehs at gmail.com (HS)
Date: Mon, 22 Jul 2024 15:55:35 +0500
Subject: [OpenSIPS-Users] No Redirection to VM on Busy/Decline
In-Reply-To: <CA+PgBobwGy=JRxKRknvPM7CKdCbwFCqoYV7BURok487iC+jpEg@mail.gmail.com>
References: <CA+PgBobwGy=JRxKRknvPM7CKdCbwFCqoYV7BURok487iC+jpEg@mail.gmail.com>
Message-ID: <CA+PgBoZK+58cp2d6gbhn+2g_g7DaPGEXgfrHr2N-UDqt-L+6zg@mail.gmail.com>

This seems to happen due to a missing line - revert_uri(); the failure
route snippet needs to be:

if (t_check_status("487|408|486|480|603")){
revert_uri();
                 rewritehostport("VM.IP:5090");
                t_relay();
                exit;


On Thu, Jul 11, 2024 at 6:54 PM HS <bullehs at gmail.com> wrote:

> Hi all,
>
> Need some collective wisdom. I am using Opensips 3.0 and have the
> following snippet for failure_route:
>
> failure_route[missed_call] {
>         if (t_was_cancelled()) {
>         rtpengine_delete();
>         exit;
>         }
>
>
>         # redirect the failed to a different VM system
>         if (t_check_status("487|408|486|480|603")){
>                  rewritehostport("VM.IP:5090");
>                 t_relay();
>                 exit;
>
>                 # do not set the missed call flag again
>         }
> }
>
> The call is redirected successfully to the VM if the user is not online.
> However, if the user is busy/declines/times out - the call is not
> redirected. I get the following error. I have searched quite a few answers,
> but don't seem to cover what I seek. Thanks for the help in advance.
>
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_conn_init: no TLS client domain found
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:core:tcp_conn_new:
> failed to do proto 3 specific init for conn 0x7f6f38860760
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the socket
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]:
> ERROR:proto_tls:proto_tls_send: connect failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:msg_send: send() to
> VM.IP:5090 for proto tls/3 failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:t_forward_nonack:
> sending request failed
> Jul 11 12:56:42 ip /usr/sbin/opensips[12592]: ERROR:tm:w_t_relay:
> t_forward_nonack failed
>
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From denys.pozniak at gmail.com  Mon Jul 22 15:53:54 2024
From: denys.pozniak at gmail.com (Denys Pozniak)
Date: Mon, 22 Jul 2024 17:53:54 +0200
Subject: [OpenSIPS-Users] How to dump all keys for cachedb_local per group?
Message-ID: <CA+x3AxohULa=pr4yxcLEKOdoLwo1AbjQr743K350XAd7X6xJ8A@mail.gmail.com>

Hello!

How can I dump the entire list of keys for a group using MI?
Maybe there is some alternative way? I just have dynamic keys tied to sip
callid.

opensips-cli -x mi cache_fetch local:shared_keys *


-- 

BR,
Denys Pozniak
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From spanda at 3clogic.com  Wed Jul 24 05:52:08 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Wed, 24 Jul 2024 11:22:08 +0530
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on
	opensips config .
Message-ID: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>

Hi All ,

I am using openisp version : 3.2

I have an opensips config on which I was listening on UDP port only and
hence using sethostport to route calls to a particular destination . like
below .

          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080");
                   route(inbound);
                    exit;
                   }

Now I have to accept a call on TLS and send that to some other destination
on UDP . I have enabled the tls module and also the dependent modules like
tls_openssl,  tls_mgm .

socket=udp:192.168.0.y:5060
socket=tls:192.168.0.y:5061
socket=tcp:192.168.0.y:5060

          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080");
                   route(inbound);
                    exit;
                   }

This above configuration is not working . I am getting "477 Send Failed "

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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From Ben.Newlin at genesys.com  Wed Jul 24 13:32:42 2024
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Wed, 24 Jul 2024 13:32:42 +0000
Subject: [OpenSIPS-Users] I need some info while setiing sethostport
	on	opensips config .
In-Reply-To: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
References: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
Message-ID: <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>

Sasmita,

You need to be using $socket_out. [1]

By default, OpenSIPS will use the receiving socket as the sending socket. This means if you receive the message on TLS and do not change $socket_out then the message will be sent out the TLS socket.

Additionally, unless you are using B2BUA or maybe topology_hiding you will likely need to “remember” this protocol transition by adding a double Record-Route [2] to the message. You may also need to change any “transport” params that exist in the Request-URI and possibly the Contact header.

[1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86
[2] https://opensips.org/docs/modules/3.2.x/rr.html

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Sasmita Panda <spanda at 3clogic.com>
Date: Wednesday, July 24, 2024 at 1:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on opensips config .
 EXTERNAL EMAIL - Please use caution with links and attachments

________________________________
Hi All ,

I am using openisp version : 3.2

I have an opensips config on which I was listening on UDP port only and hence using sethostport to route calls to a particular destination . like below .

          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080<http://freeswitch-test.xyz.com:6080>");
                   route(inbound);
                    exit;
                   }

Now I have to accept a call on TLS and send that to some other destination on UDP . I have enabled the tls module and also the dependent modules like tls_openssl,  tls_mgm .

socket=udp:192.168.0.y:5060
socket=tls:192.168.0.y:5061
socket=tcp:192.168.0.y:5060
          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080<http://freeswitch-test.xyz.com:6080>");
                   route(inbound);
                    exit;
                   }

This above configuration is not working . I am getting "477 Send Failed "

Thanks & Regards
Sasmita Panda
Senior Network Testing and Software Engineer
3CLogic , ph:07827611765
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From slackway2me at gmail.com  Wed Jul 24 15:55:58 2024
From: slackway2me at gmail.com (Alexey)
Date: Wed, 24 Jul 2024 20:55:58 +0500
Subject: [OpenSIPS-Users] How to dump all keys for cachedb_local per
	group?
In-Reply-To: <CA+x3AxohULa=pr4yxcLEKOdoLwo1AbjQr743K350XAd7X6xJ8A@mail.gmail.com>
References: <CA+x3AxohULa=pr4yxcLEKOdoLwo1AbjQr743K350XAd7X6xJ8A@mail.gmail.com>
Message-ID: <CAPc6LUH+mYFAmOTOD8AGHN5FvQmPH99XRE3T+8bfXnv72zW+6Q@mail.gmail.com>

Hello,

I think 'cache_fetch_chunk' [1] MI command is exactly what you need.
But according to the documentation, it appeared only from v.3.5.

[1] https://opensips.org/docs/modules/3.5.x/cachedb_local.html#mi_cache_fetch_chunk

-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From denys.pozniak at gmail.com  Thu Jul 25 07:35:17 2024
From: denys.pozniak at gmail.com (Denys Pozniak)
Date: Thu, 25 Jul 2024 09:35:17 +0200
Subject: [OpenSIPS-Users] How to dump all keys for cachedb_local per
	group?
In-Reply-To: <CAPc6LUH+mYFAmOTOD8AGHN5FvQmPH99XRE3T+8bfXnv72zW+6Q@mail.gmail.com>
References: <CA+x3AxohULa=pr4yxcLEKOdoLwo1AbjQr743K350XAd7X6xJ8A@mail.gmail.com>
 <CAPc6LUH+mYFAmOTOD8AGHN5FvQmPH99XRE3T+8bfXnv72zW+6Q@mail.gmail.com>
Message-ID: <CA+x3Axqe9YdNWXJgThJ7ByQ8S_L9V6wCOZK3KDw=_vZe1jn82g@mail.gmail.com>

Thank you!
Yes, everything looks like I need it.

Is there anything for the versions below?


ср, 24 июл. 2024 г. в 18:00, Alexey <slackway2me at gmail.com>:

> Hello,
>
> I think 'cache_fetch_chunk' [1] MI command is exactly what you need.
> But according to the documentation, it appeared only from v.3.5.
>
> [1]
> https://opensips.org/docs/modules/3.5.x/cachedb_local.html#mi_cache_fetch_chunk
>
> --
> best regards, Alexey
> https://alexeyka.zantsev.com/
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 

BR,
Denys Pozniak
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From gerrit.jacobsen at googlemail.com  Thu Jul 25 08:44:32 2024
From: gerrit.jacobsen at googlemail.com (Gerrit Jacobsen)
Date: Thu, 25 Jul 2024 10:44:32 +0200
Subject: [OpenSIPS-Users] Job: System Engineer VoIP - Germany / HO
Message-ID: <3458194D-6A38-4E45-AEC8-E5E1D6FF4860@googlemail.com>

VIER is looking for a System Engineer VoIP

What VIER does

VIER is a global contact center technology provider. VIER develops and runs its own technology in several data centers. You will be part of the Real-time Media Processing Team which is responsible for the telephony connections between carriers, applications and call-center agents.

Your mission

You are responsible for the support of the VoIP systems as well as the associated infrastructure
Since you have sovereignty over the VoIP systems, you are also responsible for performing system upgrades and configurations.
You have a lot of freedom to design, plan, and implement new VoIP system components for internal and external projects.
You manage and implement change requests and are a specialist for special topics in the communication area.
You show a high-sense of responsibility and foresight to keep systems running 24/7
You are prepared to work occasionally also in the evening hours to roll out new features

Your skills

You have several years experience in the day-to-day operation of telecom systems
You have worked extensively with opensource VoIP software such as Kamailio / OpenSIPS, FreeSWITCH
You can administer Linux systems such as RockyLinux and Debian
You know SIP, RTP, RTCP, TLS, WebRTC protocols and can debug problems with Wireshark
You have also worked with MySQL, Git
You are communicative and like to work with team colleagues

Further beneficial skills

Experience with Ansible or Puppet for automation and configuration management.
CheckMK, Homer, Hepic, Redis, CGRates, CDR Tools, Python
Own VoIP software projects
German language skills

Please apply here: 
https://vier.jobs.personio.de/job/1638419?language=en&display=en

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From stefan.mititelu at govoip.ro  Thu Jul 25 11:34:06 2024
From: stefan.mititelu at govoip.ro (Stefan Mititelu)
Date: Thu, 25 Jul 2024 14:34:06 +0300
Subject: [OpenSIPS-Users] [SIPREC] Send re-INVITE for exising siprec session
Message-ID: <73504dd7-da14-4d32-9e2a-318294eacbcc@govoip.ro>

Hello,

I've been trying the opensips siprec module (+rtpengine) with an SRS 
server and is working pretty nice so far.

I have this setup in which multiple phones call into a conference and I 
want to ask if the following is implemented on the siprec module side:

1. First user calls into conference
2. Send INVITE for SRS (2 media ports)

3. Second user calls into conference
4. Send re-INVITE for SRS with the updated media ports (4 media ports)

Thank you,
Stefan



From slackway2me at gmail.com  Thu Jul 25 11:39:28 2024
From: slackway2me at gmail.com (Alexey)
Date: Thu, 25 Jul 2024 16:39:28 +0500
Subject: [OpenSIPS-Users] How to dump all keys for cachedb_local per
	group?
In-Reply-To: <CA+x3Axqe9YdNWXJgThJ7ByQ8S_L9V6wCOZK3KDw=_vZe1jn82g@mail.gmail.com>
References: <CA+x3AxohULa=pr4yxcLEKOdoLwo1AbjQr743K350XAd7X6xJ8A@mail.gmail.com>
 <CAPc6LUH+mYFAmOTOD8AGHN5FvQmPH99XRE3T+8bfXnv72zW+6Q@mail.gmail.com>
 <CA+x3Axqe9YdNWXJgThJ7ByQ8S_L9V6wCOZK3KDw=_vZe1jn82g@mail.gmail.com>
Message-ID: <CAPc6LUF50jp23mosKD9oksGTa5tevSJV6yAiN1tOTfMW5udMhA@mail.gmail.com>

According to the documentation, this MI function is available since v.3.5.
Maybe the developers know something more, but as for me -
I'm absolutely sure there's nothing like this for the versions below.

-- 
best regards, Alexey
https://alexeyka.zantsev.com/


From razvan at opensips.org  Thu Jul 25 14:34:52 2024
From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=)
Date: Thu, 25 Jul 2024 17:34:52 +0300
Subject: [OpenSIPS-Users] [SIPREC] Send re-INVITE for exising siprec
 session
In-Reply-To: <73504dd7-da14-4d32-9e2a-318294eacbcc@govoip.ro>
References: <73504dd7-da14-4d32-9e2a-318294eacbcc@govoip.ro>
Message-ID: <ca46e4ac-58de-4eee-9788-83d043d16023@opensips.org>

Hi, Stefan!

Unfortunately conference calls are not yet supported. Although the code 
has been designed with this idea in mind, the number of participants is 
hardcoded to 2, and there is no way to push more participants/calls into 
an existing SIPREC session.

Best regards,

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

On 7/25/24 2:34 PM, Stefan Mititelu wrote:
> Hello,
> 
> I've been trying the opensips siprec module (+rtpengine) with an SRS 
> server and is working pretty nice so far.
> 
> I have this setup in which multiple phones call into a conference and I 
> want to ask if the following is implemented on the siprec module side:
> 
> 1. First user calls into conference
> 2. Send INVITE for SRS (2 media ports)
> 
> 3. Second user calls into conference
> 4. Send re-INVITE for SRS with the updated media ports (4 media ports)
> 
> Thank you,
> Stefan
> 
> 
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


From bogdan at opensips.org  Thu Jul 25 15:02:43 2024
From: bogdan at opensips.org (Bogdan-Andrei Iancu)
Date: Thu, 25 Jul 2024 18:02:43 +0300
Subject: [OpenSIPS-Users] OpenSIPS 3.5.0 goes stable
Message-ID: <36360d33-9932-400b-a5d4-0deb2dd76ebc@opensips.org>

<https://opensips.org/About/Version-Overview-3-5-0>



      OpenSIPS 3.5.0


      goes from beta to stable



*It got stable!*

There were two full months of work, of testing, of reporting and of 
fixing, but we did it! The *OpenSIPS 3.5* release passed all the tests 
and exams and now it is labelled as a stable release, the new flagship 
of the OpenSIPS project.


Download it now <https://www.opensips.org/Downloads/Downloads>

*3.5 Philosophy*

The OpenSIPS 3.5 delivers on the *IMS (IP Multimedia Subsystem)* topic, 
addressing at this first stage, *the CSCF components, together with its 
interfaces*. But not limited to IMS, many other areas were covered in 
3.5. So key features :

  * IMS CSCF (AKA, DIAMETER, IPSEC, Presence)
  * Launch Darkly support
  * enhanced SQL operations
  * enhanced SIPREC support

Read more on 3.5
<https://www.opensips.org/About/Version-Overview-3-5-0>


Do you want to learn more on OpenSIPS 3.5?

Join us for the firsts 3.5 *OpenSIPS eBootcamp training* 
<https://opensips.org/training/OpenSIPS_eBootcamp_2024/>, for ten days 
(40 hours) intensive and practical training, covering installation, 
configuration and administration on OpenSIPS.

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

Any questions? do not hesitate to contact us <mailto:team at opensips.org>!

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


-- 
Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com
   https://www.siphub.com
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From spanda at 3clogic.com  Fri Jul 26 11:25:41 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Fri, 26 Jul 2024 16:55:41 +0530
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on
 opensips config .
In-Reply-To: <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>
References: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
 <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>
Message-ID: <CALf-Q445uymGUCsiU8gep-dOnQBUqijQNsEZLCvL044unsdxrA@mail.gmail.com>

Hi , Thanks for the reply .
Rather than configuration change , I have done through  dynamic routing .

 In the dr_gateway table I have added socket information *udp:x.x.x.x:5060*
mysql> select * from dr_gateways;
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
| id | gwid | type | address                               | strip |
pri_prefix | attrs | probe_mode | state | socket                |
description |
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
|  1 | gw4  |    3 | fs.3c.com:6080 |     0 | NULL       | NULL  |
 0 |     0 | udp:192.168.0.69:5060 | NULL        |
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+

I only have a single Invite on a single session on this leg . There is no
Re-Invite at all  . Will I need to set record_route_preset on this case as
well ?

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


On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin <Ben.Newlin at genesys.com> wrote:

> Sasmita,
>
>
>
> You need to be using $socket_out. [1]
>
>
>
> By default, OpenSIPS will use the receiving socket as the sending socket.
> This means if you receive the message on TLS and do not change $socket_out
> then the message will be sent out the TLS socket.
>
>
>
> Additionally, unless you are using B2BUA or maybe topology_hiding you will
> likely need to “remember” this protocol transition by adding a double
> Record-Route [2] to the message. You may also need to change any
> “transport” params that exist in the Request-URI and possibly the Contact
> header.
>
>
>
> [1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86
>
> [2] https://opensips.org/docs/modules/3.2.x/rr.html
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Sasmita
> Panda <spanda at 3clogic.com>
> *Date: *Wednesday, July 24, 2024 at 1:54 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] I need some info while setiing sethostport on
> opensips config .
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> ------------------------------
>
> Hi All ,
>
>
>
> I am using openisp version : 3.2
>
>
>
> I have an opensips config on which I was listening on UDP port only and
> hence using sethostport to route calls to a particular destination . like
> below .
>
>
>
>           if(is_from_gw() || ($rp=~"5505"))
>                    {
>                   sethostport("freeswitch-test.xyz.com:6080");
>                    route(inbound);
>                     exit;
>                    }
>
>
>
> Now I have to accept a call on TLS and send that to some other destination
> on UDP . I have enabled the tls module and also the dependent modules like
> tls_openssl,  tls_mgm .
>
>
>
> socket=udp:192.168.0.y:5060
> socket=tls:192.168.0.y:5061
> socket=tcp:192.168.0.y:5060
>
>           if(is_from_gw() || ($rp=~"5505"))
>                    {
>                   sethostport("freeswitch-test.xyz.com:6080");
>                    route(inbound);
>                     exit;
>                    }
>
>
>
> This above configuration is not working . I am getting "477 Send Failed "
>
>
>
> *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 alexanderhenryperkins at gmail.com  Sat Jul 27 10:19:21 2024
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Sat, 27 Jul 2024 06:19:21 -0400
Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes
Message-ID: <CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g@mail.gmail.com>

Hi All.  We have an interesting situation.  If we send calls to our
OpenSIPS proxy, they will disconnect after 600 seconds.  However, if we
send them directly to our switch, this does not happen. Is there a timer,
or something, I should be aware of?  If so, can someone point me in the
right direction?  Any help would be appreciated.  I've been trying to
figure this out for around three months.

Thank you,
Alex
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From medeanwz at gmail.com  Sat Jul 27 10:25:47 2024
From: medeanwz at gmail.com (M S)
Date: Sat, 27 Jul 2024 12:25:47 +0200
Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes
In-Reply-To: <CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g@mail.gmail.com>
References: <CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g@mail.gmail.com>
Message-ID: <CAEZrhoPb62oyd=YN4mE+3cE+9ihUsiOTYp-28xXjiecV0sNG6A@mail.gmail.com>

It can be Session-Expires header, or reinvites (in-dialog invite being sent
after 600 seconds)

On Sat, Jul 27, 2024 at 12:22 PM Alexander Perkins <
alexanderhenryperkins at gmail.com> wrote:

> Hi All.  We have an interesting situation.  If we send calls to our
> OpenSIPS proxy, they will disconnect after 600 seconds.  However, if we
> send them directly to our switch, this does not happen. Is there a timer,
> or something, I should be aware of?  If so, can someone point me in the
> right direction?  Any help would be appreciated.  I've been trying to
> figure this out for around three months.
>
> 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 alexanderhenryperkins at gmail.com  Sat Jul 27 17:33:40 2024
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Sat, 27 Jul 2024 13:33:40 -0400
Subject: [OpenSIPS-Users] Users Digest, Vol 192, Issue 26
In-Reply-To: <mailman.1.1722081601.12054.users@lists.opensips.org>
References: <mailman.1.1722081601.12054.users@lists.opensips.org>
Message-ID: <CALLkTp0knNmpybsVvf8Rh2UQikMbU1XpOFFoOfsY6gTs_ktXtw@mail.gmail.com>

Thank you!  Is there a way to extend that timer?  Or a way to catch it in a
block and ignore it?  Basically, I’m trying to figure out the best approach
to prevent that from happening.

Thank you,
Alex

On Sat, Jul 27, 2024 at 08:01 <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. Calls Disconnecting after 10 Minutes (Alexander Perkins)
>    2. Re: Calls Disconnecting after 10 Minutes (M S)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sat, 27 Jul 2024 06:19:21 -0400
> From: Alexander Perkins <alexanderhenryperkins at gmail.com>
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Subject: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes
> Message-ID:
>         <
> CALLkTp0xAb9U9XVdnb3oVBooxNMGt9U3rg5fns1sBqOP-SGM_g at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi All.  We have an interesting situation.  If we send calls to our
> OpenSIPS proxy, they will disconnect after 600 seconds.  However, if we
> send them directly to our switch, this does not happen. Is there a timer,
> or something, I should be aware of?  If so, can someone point me in the
> right direction?  Any help would be appreciated.  I've been trying to
> figure this out for around three months.
>
> Thank you,
> Alex
> -------------- next part --------------
> An HTML attachment was scrubbed...
> URL: <
> http://lists.opensips.org/pipermail/users/attachments/20240727/7613a46e/attachment-0001.html
> >
>
> ------------------------------
>
> Message: 2
> Date: Sat, 27 Jul 2024 12:25:47 +0200
> From: M S <medeanwz at gmail.com>
> To: OpenSIPS users mailling list <users at lists.opensips.org>
> Subject: Re: [OpenSIPS-Users] Calls Disconnecting after 10 Minutes
> Message-ID:
>         <CAEZrhoPb62oyd=
> YN4mE+3cE+9ihUsiOTYp-28xXjiecV0sNG6A at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> It can be Session-Expires header, or reinvites (in-dialog invite being sent
> after 600 seconds)
>
> On Sat, Jul 27, 2024 at 12:22 PM Alexander Perkins <
> alexanderhenryperkins at gmail.com> wrote:
>
> > Hi All.  We have an interesting situation.  If we send calls to our
> > OpenSIPS proxy, they will disconnect after 600 seconds.  However, if we
> > send them directly to our switch, this does not happen. Is there a timer,
> > or something, I should be aware of?  If so, can someone point me in the
> > right direction?  Any help would be appreciated.  I've been trying to
> > figure this out for around three months.
> >
> > Thank you,
> > Alex
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> >
> -------------- next part --------------
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> http://lists.opensips.org/pipermail/users/attachments/20240727/75945c69/attachment-0001.html
> >
>
> ------------------------------
>
> 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 192, Issue 26
> **************************************
>
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From spanda at 3clogic.com  Mon Jul 29 09:50:08 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 29 Jul 2024 15:20:08 +0530
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on
 opensips config .
In-Reply-To: <CALf-Q445uymGUCsiU8gep-dOnQBUqijQNsEZLCvL044unsdxrA@mail.gmail.com>
References: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
 <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>
 <CALf-Q445uymGUCsiU8gep-dOnQBUqijQNsEZLCvL044unsdxrA@mail.gmail.com>
Message-ID: <CALf-Q47c6g=rfqPvuSdH5wC22yZBCsoyuHrJxzrBhrGn0r4iRw@mail.gmail.com>

Any suggestions on this ?



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


On Fri, Jul 26, 2024 at 4:55 PM Sasmita Panda <spanda at 3clogic.com> wrote:

> Hi , Thanks for the reply .
> Rather than configuration change , I have done through  dynamic routing .
>
>  In the dr_gateway table I have added socket information
> *udp:x.x.x.x:5060*
> mysql> select * from dr_gateways;
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
> | id | gwid | type | address                               | strip |
> pri_prefix | attrs | probe_mode | state | socket                |
> description |
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
> |  1 | gw4  |    3 | fs.3c.com:6080 |     0 | NULL       | NULL  |
>    0 |     0 | udp:192.168.0.69:5060 | NULL        |
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
>
> I only have a single Invite on a single session on this leg . There is no
> Re-Invite at all  . Will I need to set record_route_preset on this case as
> well ?
>
> *Thanks & Regards*
> *Sasmita Panda*
> *Senior Network Testing and Software Engineer*
> *3CLogic , ph:07827611765*
>
>
> On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin <Ben.Newlin at genesys.com> wrote:
>
>> Sasmita,
>>
>>
>>
>> You need to be using $socket_out. [1]
>>
>>
>>
>> By default, OpenSIPS will use the receiving socket as the sending socket.
>> This means if you receive the message on TLS and do not change $socket_out
>> then the message will be sent out the TLS socket.
>>
>>
>>
>> Additionally, unless you are using B2BUA or maybe topology_hiding you
>> will likely need to “remember” this protocol transition by adding a double
>> Record-Route [2] to the message. You may also need to change any
>> “transport” params that exist in the Request-URI and possibly the Contact
>> header.
>>
>>
>>
>> [1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86
>>
>> [2] https://opensips.org/docs/modules/3.2.x/rr.html
>>
>>
>>
>> Ben Newlin
>>
>>
>>
>> *From: *Users <users-bounces at lists.opensips.org> on behalf of Sasmita
>> Panda <spanda at 3clogic.com>
>> *Date: *Wednesday, July 24, 2024 at 1:54 AM
>> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
>> *Subject: *[OpenSIPS-Users] I need some info while setiing sethostport
>> on opensips config .
>>
>> * EXTERNAL EMAIL - Please use caution with links and attachments *
>>
>>
>> ------------------------------
>>
>> Hi All ,
>>
>>
>>
>> I am using openisp version : 3.2
>>
>>
>>
>> I have an opensips config on which I was listening on UDP port only and
>> hence using sethostport to route calls to a particular destination . like
>> below .
>>
>>
>>
>>           if(is_from_gw() || ($rp=~"5505"))
>>                    {
>>                   sethostport("freeswitch-test.xyz.com:6080");
>>                    route(inbound);
>>                     exit;
>>                    }
>>
>>
>>
>> Now I have to accept a call on TLS and send that to some other
>> destination on UDP . I have enabled the tls module and also the dependent
>> modules like tls_openssl,  tls_mgm .
>>
>>
>>
>> socket=udp:192.168.0.y:5060
>> socket=tls:192.168.0.y:5061
>> socket=tcp:192.168.0.y:5060
>>
>>           if(is_from_gw() || ($rp=~"5505"))
>>                    {
>>                   sethostport("freeswitch-test.xyz.com:6080");
>>                    route(inbound);
>>                     exit;
>>                    }
>>
>>
>>
>> This above configuration is not working . I am getting "477 Send Failed "
>>
>>
>>
>> *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 Ben.Newlin at genesys.com  Mon Jul 29 13:05:57 2024
From: Ben.Newlin at genesys.com (Ben Newlin)
Date: Mon, 29 Jul 2024 13:05:57 +0000
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on
 opensips config .
In-Reply-To: <CALf-Q47c6g=rfqPvuSdH5wC22yZBCsoyuHrJxzrBhrGn0r4iRw@mail.gmail.com>
References: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
 <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>
 <CALf-Q445uymGUCsiU8gep-dOnQBUqijQNsEZLCvL044unsdxrA@mail.gmail.com>
 <CALf-Q47c6g=rfqPvuSdH5wC22yZBCsoyuHrJxzrBhrGn0r4iRw@mail.gmail.com>
Message-ID: <BN0PR10MB5208D1DE65B1DD2D5AFB599295B72@BN0PR10MB5208.namprd10.prod.outlook.com>

Unfortunately, I have not used the database provisioning method, so I don’t have much help there. I’ll have to defer to someone else. The only thing I could think is to make sure the IP address in the DB corresponds to the socket definition on the box, but it seems like you’ve already done that.

Whether or not you need to Record-Route the protocol change depends on your local implementation. If you have dedicated or known routing channels for sequential requests – which includes ACK & BYE, not just Re-INVITE – then you may not need the routes to be in the SIP messaging. Otherwise, you probably do need the Record-Route.

Ben Newlin

From: Users <users-bounces at lists.opensips.org> on behalf of Sasmita Panda <spanda at 3clogic.com>
Date: Monday, July 29, 2024 at 5:51 AM
To: OpenSIPS users mailling list <users at lists.opensips.org>
Subject: Re: [OpenSIPS-Users] I need some info while setiing sethostport on opensips config .
 EXTERNAL EMAIL - Please use caution with links and attachments

________________________________
Any suggestions on this ?



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


On Fri, Jul 26, 2024 at 4:55 PM Sasmita Panda <spanda at 3clogic.com<mailto:spanda at 3clogic.com>> wrote:
Hi , Thanks for the reply .
Rather than configuration change , I have done through  dynamic routing .

 In the dr_gateway table I have added socket information udp:x.x.x.x:5060
mysql> select * from dr_gateways;
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
| id | gwid | type | address                               | strip | pri_prefix | attrs | probe_mode | state | socket                | description |
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
|  1 | gw4  |    3 | fs.3c.com:6080<http://fs.3c.com:6080> |     0 | NULL       | NULL  |          0 |     0 | udp:192.168.0.69:5060<http://192.168.0.69:5060> | NULL        |
+----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+

I only have a single Invite on a single session on this leg . There is no Re-Invite at all  . Will I need to set record_route_preset on this case as well ?

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


On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin <Ben.Newlin at genesys.com<mailto:Ben.Newlin at genesys.com>> wrote:
Sasmita,

You need to be using $socket_out. [1]

By default, OpenSIPS will use the receiving socket as the sending socket. This means if you receive the message on TLS and do not change $socket_out then the message will be sent out the TLS socket.

Additionally, unless you are using B2BUA or maybe topology_hiding you will likely need to “remember” this protocol transition by adding a double Record-Route [2] to the message. You may also need to change any “transport” params that exist in the Request-URI and possibly the Contact header.

[1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86<https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86>
[2] https://opensips.org/docs/modules/3.2.x/rr.html<https://opensips.org/docs/modules/3.2.x/rr.html>

Ben Newlin

From: Users <users-bounces at lists.opensips.org<mailto:users-bounces at lists.opensips.org>> on behalf of Sasmita Panda <spanda at 3clogic.com<mailto:spanda at 3clogic.com>>
Date: Wednesday, July 24, 2024 at 1:54 AM
To: OpenSIPS users mailling list <users at lists.opensips.org<mailto:users at lists.opensips.org>>
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on opensips config .
 EXTERNAL EMAIL - Please use caution with links and attachments

________________________________
Hi All ,

I am using openisp version : 3.2

I have an opensips config on which I was listening on UDP port only and hence using sethostport to route calls to a particular destination . like below .

          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080<http://freeswitch-test.xyz.com:6080>");
                   route(inbound);
                    exit;
                   }

Now I have to accept a call on TLS and send that to some other destination on UDP . I have enabled the tls module and also the dependent modules like tls_openssl,  tls_mgm .

socket=udp:192.168.0.y:5060
socket=tls:192.168.0.y:5061
socket=tcp:192.168.0.y:5060
          if(is_from_gw() || ($rp=~"5505"))
                   {
                  sethostport("freeswitch-test.xyz.com:6080<http://freeswitch-test.xyz.com:6080>");
                   route(inbound);
                    exit;
                   }

This above configuration is not working . I am getting "477 Send Failed "

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>
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From spanda at 3clogic.com  Mon Jul 29 13:11:33 2024
From: spanda at 3clogic.com (Sasmita Panda)
Date: Mon, 29 Jul 2024 18:41:33 +0530
Subject: [OpenSIPS-Users] I need some info while setiing sethostport on
 opensips config .
In-Reply-To: <BN0PR10MB5208D1DE65B1DD2D5AFB599295B72@BN0PR10MB5208.namprd10.prod.outlook.com>
References: <CALf-Q46ajMBQYzGkHq2iTqkWWL9Aen1M+tsC-UjscSt_ju9Uxw@mail.gmail.com>
 <BN0PR10MB5208059389CE1A691427716295AA2@BN0PR10MB5208.namprd10.prod.outlook.com>
 <CALf-Q445uymGUCsiU8gep-dOnQBUqijQNsEZLCvL044unsdxrA@mail.gmail.com>
 <CALf-Q47c6g=rfqPvuSdH5wC22yZBCsoyuHrJxzrBhrGn0r4iRw@mail.gmail.com>
 <BN0PR10MB5208D1DE65B1DD2D5AFB599295B72@BN0PR10MB5208.namprd10.prod.outlook.com>
Message-ID: <CALf-Q47+uiGq-wTzV_h1_tqRXgV+vddd3u=7HkmN_hisaBxdJQ@mail.gmail.com>

Thank you Ben .

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


On Mon, Jul 29, 2024 at 6:39 PM Ben Newlin <Ben.Newlin at genesys.com> wrote:

> Unfortunately, I have not used the database provisioning method, so I
> don’t have much help there. I’ll have to defer to someone else. The only
> thing I could think is to make sure the IP address in the DB corresponds to
> the socket definition on the box, but it seems like you’ve already done
> that.
>
>
>
> Whether or not you need to Record-Route the protocol change depends on
> your local implementation. If you have dedicated or known routing channels
> for sequential requests – which includes ACK & BYE, not just Re-INVITE –
> then you may not need the routes to be in the SIP messaging. Otherwise, you
> probably do need the Record-Route.
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Sasmita
> Panda <spanda at 3clogic.com>
> *Date: *Monday, July 29, 2024 at 5:51 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *Re: [OpenSIPS-Users] I need some info while setiing
> sethostport on opensips config .
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> ------------------------------
>
> Any suggestions on this ?
>
>
>
>
>
>
> *Thanks & Regards*
>
> *Sasmita Panda*
>
> *Senior Network Testing and Software Engineer*
>
> *3CLogic , ph:07827611765*
>
>
>
>
>
> On Fri, Jul 26, 2024 at 4:55 PM Sasmita Panda <spanda at 3clogic.com> wrote:
>
> Hi , Thanks for the reply .
>
> Rather than configuration change , I have done through  dynamic routing .
>
>
>
>  In the dr_gateway table I have added socket information
> *udp:x.x.x.x:5060*
>
> mysql> select * from dr_gateways;
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
> | id | gwid | type | address                               | strip |
> pri_prefix | attrs | probe_mode | state | socket                |
> description |
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
> |  1 | gw4  |    3 | fs.3c.com:6080 |     0 | NULL       | NULL  |
>    0 |     0 | udp:192.168.0.69:5060 | NULL        |
>
> +----+------+------+---------------------------------------+-------+------------+-------+------------+-------+-----------------------+-------------+
>
>
>
> I only have a single Invite on a single session on this leg . There is no
> Re-Invite at all  . Will I need to set record_route_preset on this case as
> well ?
>
>
> *Thanks & Regards*
>
> *Sasmita Panda*
>
> *Senior Network Testing and Software Engineer*
>
> *3CLogic , ph:07827611765*
>
>
>
>
>
> On Wed, Jul 24, 2024 at 7:06 PM Ben Newlin <Ben.Newlin at genesys.com> wrote:
>
> Sasmita,
>
>
>
> You need to be using $socket_out. [1]
>
>
>
> By default, OpenSIPS will use the receiving socket as the sending socket.
> This means if you receive the message on TLS and do not change $socket_out
> then the message will be sent out the TLS socket.
>
>
>
> Additionally, unless you are using B2BUA or maybe topology_hiding you will
> likely need to “remember” this protocol transition by adding a double
> Record-Route [2] to the message. You may also need to change any
> “transport” params that exist in the Request-URI and possibly the Contact
> header.
>
>
>
> [1] https://www.opensips.org/Documentation/Script-CoreVar-3-2#toc86
>
> [2] https://opensips.org/docs/modules/3.2.x/rr.html
>
>
>
> Ben Newlin
>
>
>
> *From: *Users <users-bounces at lists.opensips.org> on behalf of Sasmita
> Panda <spanda at 3clogic.com>
> *Date: *Wednesday, July 24, 2024 at 1:54 AM
> *To: *OpenSIPS users mailling list <users at lists.opensips.org>
> *Subject: *[OpenSIPS-Users] I need some info while setiing sethostport on
> opensips config .
>
> * EXTERNAL EMAIL - Please use caution with links and attachments *
>
>
> ------------------------------
>
> Hi All ,
>
>
>
> I am using openisp version : 3.2
>
>
>
> I have an opensips config on which I was listening on UDP port only and
> hence using sethostport to route calls to a particular destination . like
> below .
>
>
>
>           if(is_from_gw() || ($rp=~"5505"))
>                    {
>                   sethostport("freeswitch-test.xyz.com:6080");
>                    route(inbound);
>                     exit;
>                    }
>
>
>
> Now I have to accept a call on TLS and send that to some other destination
> on UDP . I have enabled the tls module and also the dependent modules like
> tls_openssl,  tls_mgm .
>
>
>
> socket=udp:192.168.0.y:5060
> socket=tls:192.168.0.y:5061
> socket=tcp:192.168.0.y:5060
>
>           if(is_from_gw() || ($rp=~"5505"))
>                    {
>                   sethostport("freeswitch-test.xyz.com:6080");
>                    route(inbound);
>                     exit;
>                    }
>
>
>
> This above configuration is not working . I am getting "477 Send Failed "
>
>
>
> *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
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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From oualla.simohamed at gmail.com  Wed Jul 31 01:26:52 2024
From: oualla.simohamed at gmail.com (Mohamed OUALLA)
Date: Wed, 31 Jul 2024 02:26:52 +0100
Subject: [OpenSIPS-Users] DIfference between R-URI and Destination URI in
	openSIPs
Message-ID: <CAGmsPDt+50kcvtegRStwY+MfaRnfT9nixzrOvuXmXfhEFn5XPA@mail.gmail.com>

Hello all,

  I have a technical question about the difference between *Request URI*
 and *Destination URI* in SIP. In my understanding of SIP, the R-URI
(Request URI) is located in the start line of the SIP request and is also
known as the Address of Record (AoR). However, I am unclear about what the
Destination URI is for openSIPs. Is it the same as the Request URI, or is
it related to an added route header, or the destination address in the
transport protocol, I am not sure about it?

  Additionally, I have observed that when I change the *$du* pseudo
variable in OpenSIPS, it relays the request to the UAS without changing the
R-URI (change it with the sip uri I gave to $du pseudo variable). This
behavior is the same as using the *t_relay()* method, which also does not
change the R-URI but sends the request to the UAS. I guess that changes
have been done only for the destination address in the transport layer.

  Could someone please explain these observations and clarify the
difference between R-URI and Destination URI?
  And the best way to route calls from UAC to UAS in simple VoIP call
components (Caller - SIP Proxy - Callee), actually I change the $ru, then I
*forward()* the request stateless or *t_relay()* stateful.

Thank you.
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From abalashov at evaristesys.com  Wed Jul 31 02:55:55 2024
From: abalashov at evaristesys.com (Alex Balashov)
Date: Tue, 30 Jul 2024 22:55:55 -0400
Subject: [OpenSIPS-Users] DIfference between R-URI and Destination URI
 in openSIPs
In-Reply-To: <CAGmsPDt+50kcvtegRStwY+MfaRnfT9nixzrOvuXmXfhEFn5XPA@mail.gmail.com>
References: <CAGmsPDt+50kcvtegRStwY+MfaRnfT9nixzrOvuXmXfhEFn5XPA@mail.gmail.com>
Message-ID: <9437C142-614D-400A-9D51-0BEE9C22D0DC@evaristesys.com>

Hi,

The Request URI is a SIP concept, while the destination URI might be best described as a fictive invention of OpenSIPS configuration script. It represents the next-hop destination to which the request will be forwarded on the network and transport layer, as you correctly surmised, while the request URI is a logical destination. The destination URI supersedes the request URI, but if the destination URI is not set, the domain/port/transport attributes of the request URI are consumed to determine the forwarding destination.

An RURI is not the same thing as an Address of Record; an AoR refers to a logical URI entity in the location service (registrar) context. The purpose of a registrar is to map an AoR (such as sip:mohamed at sip.opensips.org) to one or more Contact URIs (e.g. sip:line1 at 192.168.1.100;user=phone), which indicate how to reach a given device on the network and transport layer.

Hopefully that helps!

-- Alex

> On Jul 30, 2024, at 9:26 PM, Mohamed OUALLA <oualla.simohamed at gmail.com> wrote:
> 
> Hello all,
> 
>   I have a technical question about the difference between Request URI and Destination URI in SIP. In my understanding of SIP, the R-URI (Request URI) is located in the start line of the SIP request and is also known as the Address of Record (AoR). However, I am unclear about what the Destination URI is for openSIPs. Is it the same as the Request URI, or is it related to an added route header, or the destination address in the transport protocol, I am not sure about it?
> 
>   Additionally, I have observed that when I change the $du pseudo variable in OpenSIPS, it relays the request to the UAS without changing the R-URI (change it with the sip uri I gave to $du pseudo variable). This behavior is the same as using the t_relay() method, which also does not change the R-URI but sends the request to the UAS. I guess that changes have been done only for the destination address in the transport layer.
> 
>   Could someone please explain these observations and clarify the difference between R-URI and Destination URI?
>   And the best way to route calls from UAC to UAS in simple VoIP call components (Caller - SIP Proxy - Callee), actually I change the $ru, then I forward() the request stateless or t_relay() stateful.
> 
> Thank you.
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

-- 
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800



From oualla.simohamed at gmail.com  Wed Jul 31 11:46:54 2024
From: oualla.simohamed at gmail.com (Mohamed OUALLA)
Date: Wed, 31 Jul 2024 12:46:54 +0100
Subject: [OpenSIPS-Users] DIfference between R-URI and Destination URI
	in openSIPs
In-Reply-To: <9437C142-614D-400A-9D51-0BEE9C22D0DC@evaristesys.com>
References: <CAGmsPDt+50kcvtegRStwY+MfaRnfT9nixzrOvuXmXfhEFn5XPA@mail.gmail.com>
 <9437C142-614D-400A-9D51-0BEE9C22D0DC@evaristesys.com>
Message-ID: <CAGmsPDvoKXVh55gKCEkOe_g3vbB6nSWAdrWt03thUCaievYgnA@mail.gmail.com>

Hi Alex,

  Thank you for the detailed clarification.

  I appreciate the explanation that the *Destination URI* in OpenSIPS
configuration script represents *the next-hop destination at the network
and transport layer!* while the Request URI is a logical destination in the
SIP message.

  Additionally, I have observed similar behavior in SIP tools like SIPp and
sipsak, where subsequent requests in the case of *SIPp* are sent to the
first proxy instead of directly to the UAS, even if they use the R-URI of
the UAS or callee, while analysing the addresses in the transport layer
using wireshark I have seen that these tools use different address. I was
able to resolve this issue by using the *setdest *action in SIPp's xml
scenario, ensuring that the requests are routed correctly.

  Thank you again for clarifying that the Destination URI takes routing
priority over the Request URI, similar to how the Route header works
against the R-URI.


  I also apologize for the confusion regarding the AoR and R-URI; as you
mentioned, the AoR indicates "who I am", while the Contact address
indicates "where I am", used by location & Registrar services.


  One last question: What is the most effective way to connect the caller
and callee in a simple VoIP call setup (Caller - SIP Proxy - Callee)?
Should I change the *$ru* and then use *forward()* for stateless routing or
*t_relay()* for stateful routing, or is there another method you would
recommend?

Mohamed.


On Wed, Jul 31, 2024 at 4:00 AM Alex Balashov <abalashov at evaristesys.com>
wrote:

> Hi,
>
> The Request URI is a SIP concept, while the destination URI might be best
> described as a fictive invention of OpenSIPS configuration script. It
> represents the next-hop destination to which the request will be forwarded
> on the network and transport layer, as you correctly surmised, while the
> request URI is a logical destination. The destination URI supersedes the
> request URI, but if the destination URI is not set, the
> domain/port/transport attributes of the request URI are consumed to
> determine the forwarding destination.
>
> An RURI is not the same thing as an Address of Record; an AoR refers to a
> logical URI entity in the location service (registrar) context. The purpose
> of a registrar is to map an AoR (such as sip:mohamed at sip.opensips.org) to
> one or more Contact URIs (e.g. sip:line1 at 192.168.1.100;user=phone), which
> indicate how to reach a given device on the network and transport layer.
>
> Hopefully that helps!
>
> -- Alex
>
> > On Jul 30, 2024, at 9:26 PM, Mohamed OUALLA <oualla.simohamed at gmail.com>
> wrote:
> >
> > Hello all,
> >
> >   I have a technical question about the difference between Request URI
> and Destination URI in SIP. In my understanding of SIP, the R-URI (Request
> URI) is located in the start line of the SIP request and is also known as
> the Address of Record (AoR). However, I am unclear about what the
> Destination URI is for openSIPs. Is it the same as the Request URI, or is
> it related to an added route header, or the destination address in the
> transport protocol, I am not sure about it?
> >
> >   Additionally, I have observed that when I change the $du pseudo
> variable in OpenSIPS, it relays the request to the UAS without changing the
> R-URI (change it with the sip uri I gave to $du pseudo variable). This
> behavior is the same as using the t_relay() method, which also does not
> change the R-URI but sends the request to the UAS. I guess that changes
> have been done only for the destination address in the transport layer.
> >
> >   Could someone please explain these observations and clarify the
> difference between R-URI and Destination URI?
> >   And the best way to route calls from UAC to UAS in simple VoIP call
> components (Caller - SIP Proxy - Callee), actually I change the $ru, then I
> forward() the request stateless or t_relay() stateful.
> >
> > Thank you.
> > _______________________________________________
> > Users mailing list
> > Users at lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> --
> Alex Balashov
> Principal Consultant
> Evariste Systems LLC
> Web: https://evaristesys.com
> Tel: +1-706-510-6800
>
>
> _______________________________________________
> Users mailing list
> Users at lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
==============================
Mohamed OUALLA
*VoIP Technical Solutions and Software Engineer*
Mail: oualla.simohamed at gmail.com
N.Phone: +212 6 29 19 3116
<oualla.simohamed at gmail.com>*SSC Certified Professional*
==============================
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From ksrigo at gmail.com  Wed Jul 31 14:02:26 2024
From: ksrigo at gmail.com (Srigo Kanapathipillai)
Date: Wed, 31 Jul 2024 16:02:26 +0200
Subject: [OpenSIPS-Users] Removing Identity hdr
In-Reply-To: <8e8911ce-bb96-4dd6-8728-cd4085643505@opensips.org>
References: <DB9P251MB0045AB6E0056157BE57BA2C9FCCE2@DB9P251MB0045.EURP251.PROD.OUTLOOK.COM>
 <8e8911ce-bb96-4dd6-8728-cd4085643505@opensips.org>
Message-ID: <208D56BC-35D7-4425-91D1-D97CD4A872EB@gmail.com>

Hi Bogdan, 

Thank you for the fix. I have tested with Opensips 3.4.7 and it works as expected now.

Regards,
Srigo

> On 5 Jul 2024, at 10:53, Bogdan-Andrei Iancu <bogdan at opensips.org> wrote:
> 
> Hi Srigo,
> 
> You the remove in the right way, nothing more you can do about it. The problem is how the remove works and how `stir_shaken_verify()` tests for the hdr - the two are incompatible.
> So, IMHO, we should remove from the `stir_shaken_verify()` function the check on the Identity hdr presence . I just pushed this fix on 3.4/3.5/master versions.
> 
> Regards,
> 
> Bogdan-Andrei Iancu
> 
> OpenSIPS Founder and Developer
>  https://www.opensips-solutions.com
>  https://www.siphub.com
> 
> On 18.06.2024 09:31, Srigo Kanapathipillai wrote:
>> Hi,
>> 
>> I'm encountering an issue with removing an Identity header in OpenSIPS 3.4. Here’s the situation:
>> 
>> 1. An incoming call with an Identity header is received.
>> 2. I perform a `stir_shaken_verify()` and remove the Identity header in a request route.
>> 3. The call is forwarded to an upstream server, but it fails.
>> 4. In the `failure_route`, I need to forward the call to a PSTN number.
>> 
>> 5. Before sending the call to the PSTN (in compliance with French STIR/SHAKEN regulations), I need to sign it with my certificate.
>> 
>> However, when I call `stir_shaken_auth()`, I receive an error -2 indicating that the Identity header already exists. Despite running `remove_hf(identity)` before calling this function, the header isn't removed, and `$hdr(identity)` still returns the initial value of the Identity header.
>> 
>> What is the best way to remove the existing Identity header and re-sign the call?
>> 
>> Thank you,
>> Srigo
>> 
>> _______________________________________________
>> Users mailing list
>> Users at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> 



From alexanderhenryperkins at gmail.com  Wed Jul 31 15:49:55 2024
From: alexanderhenryperkins at gmail.com (Alexander Perkins)
Date: Wed, 31 Jul 2024 11:49:55 -0400
Subject: [OpenSIPS-Users] DTMF Issue
Message-ID: <CALLkTp1ZAJ-Gbxvji-PiEEGhsfWqCbq3fMzATFqLZ9Xu+eZeTQ@mail.gmail.com>

Hi All.  We have an issue where when we send a call to our client, they say
the DTMF is not working and we've confirmed that to be the case.  Our
carrier is sending over RFC2833.  How can we set the DTMF mode in
OpenSIPS?  What is the recommended approach?

Thank you,
Alex
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