[OpenSIPS-Users] OpenSIPS 3.5.0 goes stable

2024-07-25 Thread Bogdan-Andrei Iancu

<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:t...@opensips.org>!

--------


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  https://www.siphub.com
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] OpenSIPS 3.5.0 major release, beta version

2024-07-19 Thread Bogdan-Andrei Iancu

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

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[OpenSIPS-Users] OpenSIPS Bootcamp training 2024

2024-07-08 Thread Bogdan-Andrei Iancu

<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:bootc...@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-le...@lists.opensips.org?Subject=Unsubscribe>.



--
  Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] I need some help on opensips logging behaviour .

2024-07-05 Thread Bogdan-Andrei Iancu
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 
 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/

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Re: [OpenSIPS-Users] Compiling modules

2024-07-05 Thread Bogdan-Andrei Iancu

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

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Re: [OpenSIPS-Users] Removing Identity hdr

2024-07-05 Thread Bogdan-Andrei Iancu

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

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Re: [OpenSIPS-Users] BYE CDR Question

2024-07-05 Thread Bogdan-Andrei Iancu

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

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Re: [OpenSIPS-Users] I need help on opensips-trap and opensips-dbg package for debugging .

2024-07-05 Thread Bogdan-Andrei Iancu

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 .



*/
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/Sasmita Panda/
/Senior Network Testing and Software Engineer/
/3CLogic , ph:07827611765/

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Re: [OpenSIPS-Users] I need some help on opensips logging behaviour .

2024-07-05 Thread Bogdan-Andrei Iancu

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/

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Re: [OpenSIPS-Users] b2b_logic_xml compilation Error

2024-06-11 Thread Bogdan-Andrei Iancu
Indeed, there was a leftover in the Makefile.conf.template, where the 
module was still listed for exclusion. I fixed this. To take effect on 
your setup, do a git pull and delete your local Makefile.conf file.


Regards,

Bogdan-Andrei Iancu

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

On 11.06.2024 10:38, Social Boh wrote:


Thank you.

the module still present in the list of modules using make menuconfig

Regards

---
I'm SoCIaL, MayBe
El 11/06/2024 a las 2:35 a. m., Bogdan-Andrei Iancu escribió:
The b2b_logic_xml is an older and deprecated modules, replaced by 
b2b_logic.


That module exists only in 3.2. For 3.4 use b2b_logic

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 11.06.2024 09:57, Social Boh wrote:


Hello,

make[1]: Entering directory '/usr/src/opensips_3_4/modules/b2b_logic'
Compiling b2bl_db.c
Compiling b2b_logic.c
Compiling b2b_logic_ctx.c
Compiling bridging.c
Compiling entity_storage.c
Compiling logic.c
Compiling records.c
Linking b2b_logic.so
make[1]: Leaving directory '/usr/src/opensips_3_4/modules/b2b_logic'

I think the problem is:

*modules/b2b_logic_xml/b2b_logic_xml.so not compiled*

not exist any directory with this name *b2b_logic_xml* inside 
modules directory


Regards

---
I'm SoCIaL, MayBe
El 11/06/2024 a las 1:39 a. m., Bogdan-Andrei Iancu escribió:

Hi,

And if you try in the src dir:
    make modules modules=modules/b2b_logic

What do you get?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 10.06.2024 23:44, Social Boh wrote:


Hello list,

during opensips 3.4 compilation:

*ERROR: module modules/b2b_logic_xml/b2b_logic_xml.so not compiled*

and the module is not compiled and installed.

libxml2-devel library is installed.

Regards

--
---
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Re: [OpenSIPS-Users] b2b_logic_xml compilation Error

2024-06-11 Thread Bogdan-Andrei Iancu

The b2b_logic_xml is an older and deprecated modules, replaced by b2b_logic.

That module exists only in 3.2. For 3.4 use b2b_logic

Regards,

Bogdan-Andrei Iancu

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

On 11.06.2024 09:57, Social Boh wrote:


Hello,

make[1]: Entering directory '/usr/src/opensips_3_4/modules/b2b_logic'
Compiling b2bl_db.c
Compiling b2b_logic.c
Compiling b2b_logic_ctx.c
Compiling bridging.c
Compiling entity_storage.c
Compiling logic.c
Compiling records.c
Linking b2b_logic.so
make[1]: Leaving directory '/usr/src/opensips_3_4/modules/b2b_logic'

I think the problem is:

*modules/b2b_logic_xml/b2b_logic_xml.so not compiled*

not exist any directory with this name *b2b_logic_xml* inside modules 
directory


Regards

---
I'm SoCIaL, MayBe
El 11/06/2024 a las 1:39 a. m., Bogdan-Andrei Iancu escribió:

Hi,

And if you try in the src dir:
    make modules modules=modules/b2b_logic

What do you get?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 10.06.2024 23:44, Social Boh wrote:


Hello list,

during opensips 3.4 compilation:

*ERROR: module modules/b2b_logic_xml/b2b_logic_xml.so not compiled*

and the module is not compiled and installed.

libxml2-devel library is installed.

Regards

--
---
I'm SoCIaL, MayBe

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Re: [OpenSIPS-Users] b2b_logic_xml compilation Error

2024-06-11 Thread Bogdan-Andrei Iancu

Hi,

And if you try in the src dir:
    make modules modules=modules/b2b_logic

What do you get?

Regards,

Bogdan-Andrei Iancu

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

On 10.06.2024 23:44, Social Boh wrote:


Hello list,

during opensips 3.4 compilation:

*ERROR: module modules/b2b_logic_xml/b2b_logic_xml.so not compiled*

and the module is not compiled and installed.

libxml2-devel library is installed.

Regards

--
---
I'm SoCIaL, MayBe

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Re: [OpenSIPS-Users] Tracer module integration with Opensips 3.4

2024-06-11 Thread Bogdan-Andrei Iancu

Hi,

For sure it's because of the missing db_mysql module - check *ALL* the 
opensips logs, most probably the module failed to load.


Regards,

Bogdan-Andrei Iancu

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

On 07.06.2024 15:54, amel.gue...@sofrecom.com wrote:


Hello Bogdan,

Yes we load the module

"db_mysql.so"/* This is a module which provides MySQL 
connectivity for OpenSIPS *


BR, Amel

Hi,
Quick one: do you load the db_mysql.so module too ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
https://www.siphub.com

*De :*GUESMI Amel SOFRECOM 
*Envoyé :* mercredi 8 mai 2024 15:53
*À :* OpenSIPS users mailling list 
*Cc :* DL FT-FR Sbc TEAM 
*Objet :* RE: Tracer module integration with Opensips 3.4

Hello,

Any help please regarding my question ?

Thank  you

BR, Amel

*De :*GUESMI Amel SOFRECOM
*Envoyé :* lundi 29 avril 2024 10:57
*À :* OpenSIPS users mailling list 
*Cc :* DL FT-FR Sbc TEAM 
*Objet :* Tracer module integration with Opensips 3.4

Hello Everyone,

I need your support to add tracer module in order tostore 
incoming/outgoing SIP messages in database.

I already add some configs to my opensips.cfg file:

*### Tracer ###*

*loadmodule "tracer.so"*

*modparam("tracer", "trace_on", 1)*

*modparam("tracer", "trace_local_ip", "opensips:5060")*

*modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")

**….*

*if ( is_method("INVITE")) {*

*record_route();*

*do_accounting("db|log", "cdr|missed", "acc");*

*trace($var(trace_id), "d", "sip|xlog", $var(user));*

**

*  t_relay();*

*exit;*

*}*

**

The error in Opensips logs is:

ERROR:core:db_check_api: module db_mysql does not export db_use_table 
function


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:get_db_struct: 
*unable to bind database module*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] 
ERROR:tracer:parse_siptrace_id: Invalid parameters extracted!url 
! table name !


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:parse_trace_id: 
*failed to parse tracer uri []*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:Traceback (last 
included file at the bottom):


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL: 0. 
/etc/opensips/opensips.cfg


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:core:yyerror: parse 
error in /etc/opensips/opensips.cfg:53:19-20: *Parameter  
not found in module  - can't set*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_on", 1)


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_local_ip", "opensips:5060")


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")


I think that the module should store the messages in sip_trace table 
but I didn’t understand how to configure properly the trace_id with 
mysql module.

Could you help me please ?


Thank you.
Best Regards,
Amel on behalf of my colleague Chaker


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Re: [OpenSIPS-Users] E_ACC_EVENT Question

2024-06-05 Thread Bogdan-Andrei Iancu
yes, good spotting, Ben is right here, all the fields provided by an 
event must be fetched via $param()


Regards,

Bogdan-Andrei Iancu

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

On 05.06.2024 18:55, Ben Newlin wrote:


If I’m understanding correctly, acc_extra variables are only used 
during message processing; they are not visible in event routes. 
Parameters passed to event routes are accessed using the $param() 
notation [1].


Per the ACC module docs [2]:

  * /Events accounting/ - log_name will be the name of the parameter
in the event raised.

So I believe in your config it would be:

#Extra Params
        $var(src_ip) = $param(source_ip);

        $var(dst_ip) = $param (destination_ip);

Note that the param name is the log name, not the tag.

As an additional note, your reference to the $si variable as a sanity 
check is not valid as that variable is also only available in a 
message processing route. Its value is not defined in an event route.


[1] - https://www.opensips.org/Documentation/Script-Routes-3-4#toc9

[2] - https://opensips.org/docs/modules/3.4.x/acc.html#ACC-extra-id

Ben Newlin

*From: *Users  on behalf of 
Alexander Perkins 

*Date: *Wednesday, June 5, 2024 at 11:37 AM
*To: *Bogdan-Andrei Iancu 
*Cc: *OpenSIPS users mailling list 
*Subject: *Re: [OpenSIPS-Users] E_ACC_EVENT Question

* EXTERNAL EMAIL - Please use caution with links and attachments *



Hi Bogdan. Thank you for getting back to me.  When we take a look at 
what populates, they are either incorrect (for example, src_ip is 
always 0.0.0.0, but we see it correct in the route block), or they are 
null.


Let me know if this helps.

Thank you,

Alex

On Wed, Jun 5, 2024 at 11:33 AM Bogdan-Andrei Iancu 
 wrote:


Hi Alexander.

When you say "they don't seem to populate correctly", you mean
they are NULL val, or wrong value?

Regards,

    Bogdan-Andrei Iancu

OpenSIPS Founder and Developer

   https://www.opensips-solutions.com

   https://www.siphub.com

On 04.06.2024 23:59, Alexander Perkins wrote:

Hi All. We are working on implementing the E_ACC_EVENT so we
may use that for accounting purposes instead of the DB.  Now,
most things appear to work as they should.  However, when we
attempt to access the extra accounting fields, they don't seem
to populate correctly (I've highlighted the extra variables). 
Here is the config:

modparam("acc","extra_fields","evi: src_ip -> source_ip;
dst_ip -> destination_ip)

event_route [E_ACC_EVENT]
{
        #log on entry
        xlog("L_INFO","entered event_route\n");

        #Default Params
        $var(method) = $param(method);

        #Extra Params
$var(src_ip) = $acc_extra(src_ip);

        $var(source_test) = $si;

$var(dst_ip) = $acc_extra(dst_ip);

        #log vars to test
        xlog("L_INFO","heres the ER variable:
$var(source_test) and $var(src_ip)\n");

}

What am I missing? Can someone point me in the right
direction?  Any help is really appreciated.

Thank you,

Alex Perkins

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Re: [OpenSIPS-Users] E_ACC_EVENT Question

2024-06-05 Thread Bogdan-Andrei Iancu

Hi Alexander.

When you say "they don't seem to populate correctly", you mean they are 
NULL val, or wrong value?


Regards,

Bogdan-Andrei Iancu

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

On 04.06.2024 23:59, Alexander Perkins wrote:
Hi All. We are working on implementing the E_ACC_EVENT so we may use 
that for accounting purposes instead of the DB.  Now, most things 
appear to work as they should. However, when we attempt to access the 
extra accounting fields, they don't seem to populate correctly (I've 
highlighted the extra variables).  Here is the config:


modparam("acc","extra_fields","evi: src_ip -> source_ip; dst_ip -> 
destination_ip)


event_route [E_ACC_EVENT]
{
        #log on entry
        xlog("L_INFO","entered event_route\n");

        #Default Params
        $var(method) = $param(method);

        #Extra Params
$var(src_ip) = $acc_extra(src_ip);

        $var(source_test) = $si;

$var(dst_ip) = $acc_extra(dst_ip);

        #log vars to test
        xlog("L_INFO","heres the ER variable: $var(source_test) and 
$var(src_ip)\n");


}

What am I missing?  Can someone point me in the right direction?  Any 
help is really appreciated.


Thank you,
Alex Perkins

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Re: [OpenSIPS-Users] on DB read or write.

2024-06-05 Thread Bogdan-Andrei Iancu

Hi Johan,

You cannot do it like read versus write, but you can balance them, see 
the db_virtual module.


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

Regards,

Bogdan-Andrei Iancu

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

On 05.06.2024 11:12, Johan De Clercq wrote:

Hello,

is it possible to split out DB module connections (e.g. auth_db) in 
multiple connections (e.g. 1 for read and 1 for write) ?


BR, JOhan.

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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-06-05 Thread Bogdan-Andrei Iancu

Check this https://opensips.org/Documentation/TroubleShooting-Crash

Regards,

Bogdan-Andrei Iancu

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

On 04.06.2024 16:31, Sasmita Panda wrote:
In the logs it says core dumped , but I am not getting the core file 
anywhere in the system .



/usr/local/sbin/opensips 
-f /usr/local/etc/opensips/opensips_webrtc_proxy_homer.cfg

This is the command I am running  .



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


On Tue, Jun 4, 2024 at 6:42 PM Bogdan-Andrei Iancu 
 wrote:


 1. Segmentation fault (core dumped)


Could you post the backtrace ?

Regards

Bogdan-Andrei Iancu

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

On 04.06.2024 14:19, Sasmita Panda wrote:

I am just trying to debug the same thing on a different machine
of ARM64 architecture . Which I am using for the very first time
for opensips .

The process is not getting started . The logs are attached below
. I am not getting any errors as well . Just exiting with a
Segmentation fault message .

https://pastebin.mozilla.org/cabnNDYt


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


On Mon, Jun 3, 2024 at 6:24 PM Bogdan-Andrei Iancu
 wrote:

You should check the opensips logs to check for the
ERROR/CRITICAL messages that comes with the failure of
OpenSIPS to start.

Some similar to the CRITICAL you mentioned in the prev posts.

Regards,

    Bogdan-Andrei Iancu

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

On 31.05.2024 12:01, Sasmita Panda wrote:

[opensips]      this is a line I am getting in
between the services of opensips when opensips is not
listening  .

Attached the logs with mem Debug as you mentioned .

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


On Thu, May 30, 2024 at 9:21 PM Bogdan-Andrei Iancu
 wrote:

Hi,

Try to start opensips is mem debugging support - add
"|-a Q_MALLOC_DBG" to the startup cmd line of OpenSIPS -
this will give more hints on that crashing point.

Regards,
    |

        Bogdan-Andrei Iancu

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

On 27.05.2024 12:46, Sasmita Panda wrote:

Hi ,

I am just facing the same issue again .

/usr/local/sbin/opensips[3443]: CRITICAL:core:fm_free:
freeing already freed shm pointer (0x7f3a59ddb4a0),
first free: (null): (null)(0) - aborting!

whenever I am starting opensips this is the last line I
am getting in the logs and it wont listen on the
specified port .


As earlier suggested I was trying to run a trap with
opensips-cli , but I am facing an issue with that . It
says the trap module is not loaded .
How do I load the trap module of opesnips-cli
specifically ?

Please help . This is a kind of blocker for me .





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


On Thu, Mar 21, 2024 at 10:01 PM Bogdan-Andrei Iancu
 wrote:

IF

(a) it crashes, try to get a backtrace

(b) it block on starting, try to do a "trap" via
opensips-cli

        Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:24, Sasmita Panda wrote:

Sometimes it crashes and sometimes while starting
I get the warings of the timer .  Same
config  shows different issues .



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



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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-06-04 Thread Bogdan-Andrei Iancu

1. Segmentation fault (core dumped)


Could you post the backtrace ?

Regards

Bogdan-Andrei Iancu

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

On 04.06.2024 14:19, Sasmita Panda wrote:
I am just trying to debug the same thing on a different machine of 
ARM64 architecture . Which I am using for the very first time for 
opensips .


The process is not getting started . The logs are attached below . I 
am not getting any errors as well . Just exiting with a 
Segmentation fault message .


https://pastebin.mozilla.org/cabnNDYt


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


On Mon, Jun 3, 2024 at 6:24 PM Bogdan-Andrei Iancu 
 wrote:


You should check the opensips logs to check for the ERROR/CRITICAL
messages that comes with the failure of OpenSIPS to start.

Some similar to the CRITICAL you mentioned in the prev posts.

Regards,

Bogdan-Andrei Iancu

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

On 31.05.2024 12:01, Sasmita Panda wrote:

[opensips]      this is a line I am getting in
between the services of opensips when opensips is not listening  .

Attached the logs with mem Debug as you mentioned .

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


On Thu, May 30, 2024 at 9:21 PM Bogdan-Andrei Iancu
 wrote:

Hi,

Try to start opensips is mem debugging support - add "|-a
Q_MALLOC_DBG" to the startup cmd line of OpenSIPS - this will
give more hints on that crashing point.

Regards,
    |

    Bogdan-Andrei Iancu

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

On 27.05.2024 12:46, Sasmita Panda wrote:

Hi ,

I am just facing the same issue again .

/usr/local/sbin/opensips[3443]: CRITICAL:core:fm_free:
freeing already freed shm pointer (0x7f3a59ddb4a0), first
free: (null): (null)(0) - aborting!

whenever I am starting opensips this is the last line I am
getting in the logs and it wont listen on the specified port .


As earlier suggested I was trying to run a trap with
opensips-cli , but I am facing an issue with that . It says
the trap module is not loaded .
How do I load the trap module of opesnips-cli specifically ?

Please help . This is a kind of blocker for me .





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


On Thu, Mar 21, 2024 at 10:01 PM Bogdan-Andrei Iancu
 wrote:

IF

(a) it crashes, try to get a backtrace

(b) it block on starting, try to do a "trap" via
opensips-cli

        Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:24, Sasmita Panda wrote:

Sometimes it crashes and sometimes while starting I get
the warings of the timer .  Same config  shows
different issues .



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


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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-06-03 Thread Bogdan-Andrei Iancu
You should check the opensips logs to check for the ERROR/CRITICAL 
messages that comes with the failure of OpenSIPS to start.


Some similar to the CRITICAL you mentioned in the prev posts.

Regards,

Bogdan-Andrei Iancu

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

On 31.05.2024 12:01, Sasmita Panda wrote:
[opensips]      this is a line I am getting in between the 
services of opensips when opensips is not listening  .


Attached the logs with mem Debug as you mentioned .

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


On Thu, May 30, 2024 at 9:21 PM Bogdan-Andrei Iancu 
 wrote:


Hi,

Try to start opensips is mem debugging support - add "|-a
Q_MALLOC_DBG" to the startup cmd line of OpenSIPS - this will give
more hints on that crashing point.

Regards,
    |

    Bogdan-Andrei Iancu

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

On 27.05.2024 12:46, Sasmita Panda wrote:

Hi ,

I am just facing the same issue again .

/usr/local/sbin/opensips[3443]: CRITICAL:core:fm_free: freeing
already freed shm pointer (0x7f3a59ddb4a0), first free: (null):
(null)(0) - aborting!

whenever I am starting opensips this is the last line I am
getting in the logs and it wont listen on the specified port .


As earlier suggested I was trying to run a trap with opensips-cli
, but I am facing an issue with that . It says the trap module is
not loaded .
How do I load the trap module of opesnips-cli specifically ?

Please help . This is a kind of blocker for me .





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


On Thu, Mar 21, 2024 at 10:01 PM Bogdan-Andrei Iancu
 wrote:

IF

(a) it crashes, try to get a backtrace

(b) it block on starting, try to do a "trap" via opensips-cli

    Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:24, Sasmita Panda wrote:

Sometimes it crashes and sometimes while starting I get the
warings of the timer . Same config  shows different issues .



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


On Wed, Mar 20, 2024 at 6:45 PM Bogdan-Andrei Iancu
 wrote:

Hi,

How the two reports fit together here ? there are
completely separate experiences on different runs?? or
if you start opensips first you get the warnings and
later it crashes ??
For the crash part, I see a core file was generated -
could you extract the backtrace and post here ? (see
https://opensips.org/Documentation/TroubleShooting-Crash)

    Regards

Bogdan-Andrei Iancu

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

On 11.03.2024 15:04, Sasmita Panda wrote:

Any update on this ?


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


On Mon, Mar 11, 2024 at 12:03 PM Sasmita Panda
 wrote:

With the same server configuration and opensips
version I am getting below error as well .

CRITICAL:core:fm_free: freeing already freed shm
pointer (0x7fc110e0b408), first free: (null):
(null)(0) - aborting!

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Re: [OpenSIPS-Users] Opensips as proxy for Asterisk

2024-06-03 Thread Bogdan-Andrei Iancu
Ok, and where the things are getting broken with the INVITE? is an 
INVITE from the webrtc client? does it get to OpenSIPS? is OpenSIPS 
forwarding it?


Regards,

Bogdan-Andrei Iancu

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

On 02.06.2024 09:21, sterlin wrote:

Hi,
I am facing issue with INVITE .


On 31 May 2024 at 3:46 pm, Bogdan-Andrei Iancu  
wrote:


Hi,

What exact part is not working for you? The REGISTERs? or the INVITEs?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 11.04.2024 13:58, Sterlin Devanish wrote:

Hi friends,

I am new to opensips.
I am working on handling Background calls for Flutter WebRTC clients 
using Asterisk.


Since Asterisk doesn't support RFC8599, I am trying to configure 
opensips as a proxy server for Asterisk.


I am using mid_registrar to forward the registration request from 
opensips to asterisk.
It is perfectly working for SIP signaling, whereas for WebSockets 
the request is not reaching the asterisk from opensips.


Kindly help me where I am going wrong, or help me handle this scenario.

/Thanks,/
/*Sterlin Devanish D*/


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Re: [OpenSIPS-Users] Webrtc LB Opensips 3.1

2024-05-31 Thread Bogdan-Andrei Iancu

Hi,

I guess something based on DNS, considering the fact that there is 
nothing between the clients and the deployments.


Regards,

Bogdan-Andrei Iancu

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

On 29.05.2024 17:16, inderjeet sharma wrote:

Hi Team,

I'm using OpenSIPS and RTPEngine for JSSIP/WebRTC, and it's working 
fine. However, I plan to run the same setup in parallel and load 
balance the traffic between these two deployments. Could you suggest a 
solution


                   [JSSIP Client]
                          / \ / \
[OpenSIPS + RTPEngine Deployment 1] [OpenSIPS + RTPEngine Deployment 2]
                          \ / \ /
                [Asterisk Server]


Thanks
Inderjeet


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Re: [OpenSIPS-Users] opensips nat question

2024-05-31 Thread Bogdan-Andrei Iancu

Hi,

And what's the actual question ?

Regards,

Bogdan-Andrei Iancu

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

On 27.05.2024 12:18, suifeng wrote:

HI,I has an NAT question;
img one:
[...]
img two:  Use fix_nated_sdp(10) processing:
[]
opensips version OpenSIPS (3.2.17 (x86_64/linux))
opensips.cfg  part context:
route{
       if (nat_uac_test(23)) {
            if (is_method("REGISTER")) {
                setbflag("NAT");
                fix_nated_register();
                xlog("request nat: $fd, rd: $rd, ru: $ru");
                xlog("request -1");
            }
            if (is_method("INVITE")){
                xlog("request invite: $fd, rd: $rd, ru: $ru");
                fix_nated_contact();
add_rr_param(";nat=yes");
                xlog("request -2");
            }
        }

        if (is_method("INVITE") && has_body("application/sdp")) {
                xlog("request 
13-1si:[$si],cs:[$cs],uri:[uri]");

                fix_nated_sdp(10);
        }
}



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Re: [OpenSIPS-Users] Passing of PN params from Linphone to opensips

2024-05-31 Thread Bogdan-Andrei Iancu

Hi Ruben,

Could you post the at least the full Contact hdr you have?

Regards,

Bogdan-Andrei Iancu

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

On 13.05.2024 00:06, Ruben Heusel wrote:

Hey all,

i have an opensips running and followed 
https://blog.opensips.org/2020/05/07/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-i/ 
to enable Push Notification support.
I can connect to the opensips server with linphone, but the PN params 
i pass are not recognized.


DBG:core:parse_params: Parsing params 
for:[message-expires=2419200;+pn-provider="";+pn-prid="";+pn-param="";+sip.instance="";+org.linphone.specs="lime"]

DBG:core:pn_inspect_request: Contact URI has no PN params

I have no idea how to troubleshoot this and would appreciate any help.

Regards

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Re: [OpenSIPS-Users] Tracer module integration with Opensips 3.4

2024-05-31 Thread Bogdan-Andrei Iancu

Your college Chaker pushed the same question, just check my reply there ;)

Regards,

Bogdan-Andrei Iancu

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

On 08.05.2024 17:53, amel.gue...@sofrecom.com wrote:


Hello,

Any help please regarding my question ?

Thank  you

BR, Amel

*De :*GUESMI Amel SOFRECOM
*Envoyé :* lundi 29 avril 2024 10:57
*À :* OpenSIPS users mailling list 
*Cc :* DL FT-FR Sbc TEAM 
*Objet :* Tracer module integration with Opensips 3.4

Hello Everyone,

I need your support to add tracer module in order tostore 
incoming/outgoing SIP messages in database.

I already add some configs to my opensips.cfg file:

*### Tracer ###*

*loadmodule "tracer.so"*

*modparam("tracer", "trace_on", 1)*

*modparam("tracer", "trace_local_ip", "opensips:5060")*

*modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")

**….*

*if ( is_method("INVITE")) {*

*record_route();*

*do_accounting("db|log", "cdr|missed", "acc");*

*trace($var(trace_id), "d", "sip|xlog", $var(user));*

**

*  t_relay();*

*exit;*

*}*

**

The error in Opensips logs is:

ERROR:core:db_check_api: module db_mysql does not export db_use_table 
function


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:get_db_struct: 
*unable to bind database module*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] 
ERROR:tracer:parse_siptrace_id: Invalid parameters extracted!url 
! table name !


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:parse_trace_id: 
*failed to parse tracer uri []*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:Traceback (last 
included file at the bottom):


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL: 0. 
/etc/opensips/opensips.cfg


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:core:yyerror: parse 
error in /etc/opensips/opensips.cfg:53:19-20: *Parameter  
not found in module  - can't set*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_on", 1)


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_local_ip", "opensips:5060")


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")


I think that the module should store the messages in sip_trace table 
but I didn’t understand how to configure properly the trace_id with 
mysql module.

Could you help me please ?


Thank you.
Best Regards,
Amel on behalf of my colleague Chaker


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Re: [OpenSIPS-Users] route handling of http protocol in opensips

2024-05-31 Thread Bogdan-Andrei Iancu

Hi Anmol,

May I ask why do you want to handle HTTP custom traffic in OpenSIPS, 
which is a SIP server?


Best regards,

Bogdan-Andrei Iancu

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

On 01.05.2024 07:12, ANMOL PRAKASH via Users wrote:

Hi all,

Is there any module in opensips inline with xHTTP in kamailio.

In kamailio, xHTTP module offers a generic way of handling the HTTP protocol, 
by calling event_route[xhttp:request] in your config.

In opensips, httpd module is there to enable http server on opensips but I am 
looking for handling of HTTP protocol in opensips.


Any help will be highly appreciated.

Thanks & Regards

Anmol Prakash (5273)
C-DOT DELHI
INDIA

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Re: [OpenSIPS-Users] Tracer module integration with Opensips 3.4

2024-05-31 Thread Bogdan-Andrei Iancu

Hi,

Quick one: do you load the db_mysql.so module too ?

Regards,

Bogdan-Andrei Iancu

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

On 25.04.2024 11:31, chaker.barka...@sofrecom.com wrote:


Hello Everyone,

I need your support to add tracer module in order tostore 
incoming/outgoing SIP messages in database.

I already add some configs to my opensips.cfg file:

*### Tracer ###*

*loadmodule "tracer.so"*

*modparam("tracer", "trace_on", 1)*

*modparam("tracer", "trace_local_ip", "opensips:5060")*

*modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")

**….*

*if ( is_method("INVITE")) {*

*record_route();*

*do_accounting("db|log", "cdr|missed", "acc");*

*trace($var(trace_id), "d", "sip|xlog", $var(user));*

**

*  t_relay();*

*exit;*

*}*

**

The error in Opensips logs is:

ERROR:core:db_check_api: module db_mysql does not export db_use_table 
function


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:get_db_struct: 
*unable to bind database module*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] 
ERROR:tracer:parse_siptrace_id: Invalid parameters extracted!url 
! table name !


2024-04-25 09:29:10 Apr 25 10:29:10 [52] ERROR:tracer:parse_trace_id: 
*failed to parse tracer uri []*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:Traceback (last 
included file at the bottom):


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL: 0. 
/etc/opensips/opensips.cfg


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:core:yyerror: parse 
error in /etc/opensips/opensips.cfg:53:19-20: *Parameter  
not found in module  - can't set*


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_on", 1)


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_local_ip", "opensips:5060")


2024-04-25 09:29:10 Apr 25 10:29:10 [52] CRITICAL:modparam("tracer", 
"trace_id","[tid]uri=mysql://opensips:opensipsrw@ossdb/opensips;table=sip_trace;")


I think that the module should store the messages in sip_trace table 
but I didn’t understand how to configure properly the trace_id with 
mysql module.

Could you help me please ?


Thank you.
Best Regards,
Chaker BARKAOUI.

Orange Restricted


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Re: [OpenSIPS-Users] Opensips as proxy for Asterisk

2024-05-31 Thread Bogdan-Andrei Iancu

Hi,

What exact part is not working for you? The REGISTERs? or the INVITEs?

Regards,

Bogdan-Andrei Iancu

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

On 11.04.2024 13:58, Sterlin Devanish wrote:

Hi friends,

I am new to opensips.
I am working on handling Background calls for Flutter WebRTC clients 
using Asterisk.


Since Asterisk doesn't support RFC8599, I am trying to configure 
opensips as a proxy server for Asterisk.


I am using mid_registrar to forward the registration request from 
opensips to asterisk.
It is perfectly working for SIP signaling, whereas for WebSockets the 
request is not reaching the asterisk from opensips.


Kindly help me where I am going wrong, or help me handle this scenario.

/Thanks,/
/*Sterlin Devanish D*/


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Re: [OpenSIPS-Users] Long reload time for mi tls_reload for 200 tls/ssl certs

2024-05-30 Thread Bogdan-Andrei Iancu

Hi Denys,

That is really weird, 4 out of 5 traps point to the shm_malloc() 
function, trying to get a free mem check. What is the usage of the shm 
mem (use the shmem: stats class to see) ? How many times you do this 
reload? does it get slow from the first?


Regards,

Bogdan-Andrei Iancu

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

On 14.05.2024 16:47, Denys Pozniak wrote:

Hello!

I disabled logging and added some resources to the virtual machine.
On a working OpenSIPS, I reloaded the tls several times and in 
parallel ran a trap.

#opensips-cli -x mi tls_reload
#opensips-cli -x trap

If possible, please analyze it again, maybe you could find something 
interesting:

https://github.com/denyspozniak/opensips_tls_debug/tree/main

Thanks in advance!



ср, 8 мая 2024 г. в 19:59, Brett Nemeroff :

Just offering my experience here. I have, without a doubt,
noticed intensive logging brings a highly performant server to its
knees.

Disable ALL logging. Watch disk IO and confirm it's disabled. Try
it again. Just an easy thing to try.

-Brett


On Wed, May 8, 2024 at 7:17 AM Bogdan-Andrei Iancu
 wrote:

Hi,

There is only one trap, ideally you should try to get several
during the reload time.

Still, the trap you did shows opensips doing some logging
(dumping to syslog) while reloading - could you check how
intensive this logging is and eventually to try to disable it
(increase the log level of opensips lower than INFO) to see if
there is any impact?

Regards,

Bogdan-Andrei Iancu

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

On 08.05.2024 14:10, Denys Pozniak wrote:

Hello!

If possible, please check log:

https://github.com/denyspozniak/opensips_tls_debug/blob/main/gdb_opensips_20240508_115956


ср, 8 мая 2024 г. в 08:55, Bogdan-Andrei Iancu
:

Hi Denys.

That is rather weird, 250 certificates in 1 min. I assume
it is not a DB issue (considering the db_text backend),
so can you try to do multiple sequential "opensips-cli -x
trap" to try to understand what is going on ?

Regards,

        Bogdan-Andrei Iancu

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

On 02.05.2024 11:41, Denys Pozniak wrote:

Hello!

There is a task to divide a single tls/ssl letsencrypt
certificate for white labels into specific ones.
I entered about ~250 certificates into db_text, but as
it turned out, for OpenSIPS this is a rather complex
operation to load them and takes about 1 minute and a
heavy CPU load is noticeable.

I would appreciate any advice on how to avoid this.

# wc -l dbtext/tls_mgm
253 dbtext/tls_mgm

# time opensips-cli -x mi tls_reload
"OK"
real 0m52.034s
user 0m1.419s
sys 0m0.433s

# time systemctl restart opensips
real    0m58.198s
user    0m0.024s
sys     0m0.055s

# opensips -V
version: opensips 3.4.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC,
FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144,
MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 036d02961

-- 


BR,
Denys Pozniak



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-- 


BR,
Denys Pozniak




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--

BR,
Denys Pozniak


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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-05-30 Thread Bogdan-Andrei Iancu

Hi,

Try to start opensips is mem debugging support - add "|-a Q_MALLOC_DBG" 
to the startup cmd line of OpenSIPS - this will give more hints on that 
crashing point.


Regards,
|

Bogdan-Andrei Iancu

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

On 27.05.2024 12:46, Sasmita Panda wrote:

Hi ,

I am just facing the same issue again .

/usr/local/sbin/opensips[3443]: CRITICAL:core:fm_free: freeing already 
freed shm pointer (0x7f3a59ddb4a0), first free: (null): (null)(0) - 
aborting!


whenever I am starting opensips this is the last line I am getting in 
the logs and it wont listen on the specified port .



As earlier suggested I was trying to run a trap with opensips-cli , 
but I am facing an issue with that . It says the trap module is not 
loaded .

How do I load the trap module of opesnips-cli specifically ?

Please help . This is a kind of blocker for me .





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


On Thu, Mar 21, 2024 at 10:01 PM Bogdan-Andrei Iancu 
 wrote:


IF

(a) it crashes, try to get a backtrace

(b) it block on starting, try to do a "trap" via opensips-cli

    Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:24, Sasmita Panda wrote:

Sometimes it crashes and sometimes while starting I get the
warings of the timer .  Same config  shows different issues .



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


On Wed, Mar 20, 2024 at 6:45 PM Bogdan-Andrei Iancu
 wrote:

Hi,

How the two reports fit together here ? there are completely
separate experiences on different runs?? or if you start
opensips first you get the warnings and later it crashes ??
For the crash part, I see a core file was generated - could
you extract the backtrace and post here ? (see
https://opensips.org/Documentation/TroubleShooting-Crash)

    Regards

Bogdan-Andrei Iancu

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

On 11.03.2024 15:04, Sasmita Panda wrote:

Any update on this ?


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


On Mon, Mar 11, 2024 at 12:03 PM Sasmita Panda
 wrote:

With the same server configuration and opensips version
I am getting below error as well .

CRITICAL:core:fm_free: freeing already freed shm pointer
(0x7fc110e0b408), first free: (null): (null)(0) - aborting!

 /usr/local/sbin/opensips[215171]:
INFO:core:handle_sigs: child process 215185 exited by a
signal 6
 /usr/local/sbin/opensips[215171]:
INFO:core:handle_sigs: core was generated
 /usr/local/sbin/opensips[215171]:
INFO:core:handle_sigs: terminating due to SIGCHLD
 /usr/local/sbin/opensips[215172]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: fm_status
(0x7fc2907ff010):
 /usr/local/sbin/opensips[215174]: INFO:core:sig_usr:
signal 15 received
 /usr/local/sbin/opensips[215172]: heap size= 33554432
 /usr/local/sbin/opensips[215174]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: used= 3710048,
used+overhead=3801344, free=29844384
 /usr/local/sbin/opensips[215174]: fm_status
(0x7fc2907ff010):
 /usr/local/sbin/opensips[215172]: max used (+overhead)=
3801344
 /usr/local/sbin/opensips[215174]: heap size= 33554432
 /usr/local/sbin/opensips[215172]: dumping free list:
 /usr/local/sbin/opensips[215174]: used= 229752,
used+overhead=318936, free=33324680
 /usr/local/sbin/opensips[215172]: hash =   1 fragments
no.:     1, unused: 0#012#011#011 bucket size:         8
-     8 (first         8)
 /usr/local/sbin/opensips[215174]: max used (+overhead)=
385792
 /usr/local/sbin/opensips[215174]: dumping free list:
 /usr/local/sbin/opensips[215172]: hash =  18 fragments
no.:     1, unused: 0#012#011#011 bucket size:       144
-   144 (first       144)
 /usr/local/sbin/opensips[215174]: hash =   7 fragments
no.:   139, unused: 0#012#011#011 bucket size:        56
-    56 (first        56)
 /usr/local/sbin/opensips[215172]: hash = 2059 fragments
no.:     1, unused: 0#012#011#011 bucket size:  16777216

[OpenSIPS-Users] OpenSIPS Summit 2024 - post facts

2024-05-30 Thread Bogdan-Andrei Iancu

Hello all !!

I would like to thank you all for being part of the OpenSIPS Summit 2024 
in Valencia. It was an amazing set of speakers, sponsors and 
participants - I hope you all enjoyed the event, and even more, I hope 
you found it useful in terms of getting the updates and news from the 
VoIP and RTC words.


Now that the event is behind us, let me fill in here some post facts

 * the presentations and recordings were uploaded at attached to the
   schedule / linked to the web site
   <https://www.opensips.org/events/Summit-2024Valencia/#schedules>
 * the recordings are also available on the OpenSIPS YouTube channel
   <https://www.youtube.com/playlist?list=PLMMZA6ketvKqMJSM6XiaBzKRxualdj-gg>
 * photos from the event were uploaded on the OpenSIPS Summit 2024
   Album <https://photos.app.goo.gl/AA7PLMLmXvhSCEXr7>


Enjoy !

--
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] Manipulation with the received parameter in Via for force_rport()

2024-05-30 Thread Bogdan-Andrei Iancu

Hi Denys,

Unfortunately this is not possible from script level - to pick the spot 
where the `received` will be inserted. Of course, there is all the time 
to options to put your fingers into the code :D .


Regards,

Bogdan-Andrei Iancu

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

On 28.05.2024 10:37, Denys Pozniak wrote:

Hello!

Is it possible to change where the received parameter is added to the 
Via header using force_rport()?
I need to put it after the branch parameter, since this turned out to 
be important for very ancient SIP terminals.


REGISTER sip:sip.local.net <http://sip.local.net> SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.216:5060;branch=z9hG4bK59d0cba274f66fa62e82a5a7d383b461;rport


REGISTER sip:sip.local.net <http://sip.local.net> SIP/2.0
Via: SIP/2.0/UDP 185.99.99.99:5060;branch=z9hG4bK92e8.8d908052.0;cid=1
Via: SIP/2.0/UDP 192.168.1.216:5060 
<http://192.168.1.216:5060>;*received=185.44.44.44*;branch=z9hG4bK59d0cba274f66fa62e82a5a7d383b461;rport=5060


Thanks in advance!


--

BR,
Denys Pozniak



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Re: [OpenSIPS-Users] The cause of 'max nr of branches exceeded'

2024-05-30 Thread Bogdan-Andrei Iancu

Hi Eran,

The error you get means you reached the maximum number of branches per 
transactions. Branches are created during parallel or serial forking and 
there are max 12 configured in OpenSIPS.


So, check your script, maybe you have a run-away forking creating more 
than 12 branches.


Regards,

Bogdan-Andrei Iancu

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

On 08.05.2024 20:27, Eran Leshem wrote:

Hi,
After a few calls, I am getting in the log the following error message:
ERROR:core:append_branch: max nr of branches exceeded
ERROR:core:push_branch: failed to append a branch

What can I do to prevent this?

In opensips.cfg:
udp_workers=2
open_files_limit=600

In /etc/default/opensips:
S_MEMORY=3200
P_MEMORY=200

My machine:
- Linux Mint 21.3 which is based on Ubuntu 22.04 LTS (Jammy Jellyfish),
- 16 GB memory,
- OpenSIPS version 3.4

Thanks,
Eran L.

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[OpenSIPS-Users] OpenSIPS 3.5.0 major release, beta version

2024-05-09 Thread Bogdan-Andrei Iancu

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
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Re: [OpenSIPS-Users] Long reload time for mi tls_reload for 200 tls/ssl certs

2024-05-08 Thread Bogdan-Andrei Iancu

Hi,

There is only one trap, ideally you should try to get several during the 
reload time.


Still, the trap you did shows opensips doing some logging (dumping to 
syslog) while reloading - could you check how intensive this logging is 
and eventually to try to disable it (increase the log level of opensips 
lower than INFO) to see if there is any impact?


Regards,

Bogdan-Andrei Iancu

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

On 08.05.2024 14:10, Denys Pozniak wrote:

Hello!

If possible, please check log:
https://github.com/denyspozniak/opensips_tls_debug/blob/main/gdb_opensips_20240508_115956


ср, 8 мая 2024 г. в 08:55, Bogdan-Andrei Iancu :

Hi Denys.

That is rather weird, 250 certificates in 1 min. I assume it is
not a DB issue (considering the db_text backend), so can you try
to do multiple sequential "opensips-cli -x trap" to try to
understand what is going on ?

Regards,

    Bogdan-Andrei Iancu

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

On 02.05.2024 11:41, Denys Pozniak wrote:

Hello!

There is a task to divide a single tls/ssl letsencrypt
certificate for white labels into specific ones.
I entered about ~250 certificates into db_text, but as it turned
out, for OpenSIPS this is a rather complex operation to load them
and takes about 1 minute and a heavy CPU load is noticeable.

I would appreciate any advice on how to avoid this.

# wc -l dbtext/tls_mgm
253 dbtext/tls_mgm

# time opensips-cli -x mi tls_reload
"OK"
real 0m52.034s
user 0m1.419s
sys 0m0.433s

# time systemctl restart opensips
real    0m58.198s
user    0m0.024s
sys     0m0.055s

# opensips -V
version: opensips 3.4.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
16, MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: 036d02961

-- 


BR,
Denys Pozniak



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--

BR,
Denys Pozniak


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Re: [OpenSIPS-Users] Long reload time for mi tls_reload for 200 tls/ssl certs

2024-05-08 Thread Bogdan-Andrei Iancu

Hi Denys.

That is rather weird, 250 certificates in 1 min. I assume it is not a DB 
issue (considering the db_text backend), so can you try to do multiple 
sequential "opensips-cli -x trap" to try to understand what is going on ?


Regards,

Bogdan-Andrei Iancu

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

On 02.05.2024 11:41, Denys Pozniak wrote:

Hello!

There is a task to divide a single tls/ssl letsencrypt certificate for 
white labels into specific ones.
I entered about ~250 certificates into db_text, but as it turned out, 
for OpenSIPS this is a rather complex operation to load them and takes 
about 1 minute and a heavy CPU load is noticeable.


I would appreciate any advice on how to avoid this.

# wc -l dbtext/tls_mgm
253 dbtext/tls_mgm

# time opensips-cli -x mi tls_reload
"OK"
real 0m52.034s
user 0m1.419s
sys 0m0.433s

# time systemctl restart opensips
real    0m58.198s
user    0m0.024s
sys     0m0.055s

# opensips -V
version: opensips 3.4.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, 
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535

poll method support: poll, epoll, sigio_rt, select.
git revision: 036d02961

--

BR,
Denys Pozniak



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[OpenSIPS-Users] OpenSIPS Summit 2024 - This is our content

2024-04-24 Thread Bogdan-Andrei Iancu


   OpenSIPS Summit, 14-17 May, 2024, Valencia, Spain

 * 2 days of conference on SIP, VoIP, RTC and Open Source
 * 1 day of demos
 * 1 day of advanced training
 * 1 cozy dinner event
 * 1 bold catamaran sea trip

<https://www.opensips.org/events/Summit-2024Valencia/>

For such a great content let's boost the opportunity with the 
*SUMMIT-24-LABOR-DAY* /50% discount code/ between *1-5 of May* - this is 
a truly great deal !


Register now <https://www.opensips.org/events/Summit-2024Valencia/#pricing>

**






    


--
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Re: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic

2024-04-24 Thread Bogdan-Andrei Iancu

Hi Julien,

The term of AS is super abused when comes to what it should deliver. 
First of all you clearly need to define what should be the 
services/functionalities you need from an "AS" and to see what solutions 
are available to implement them.


In most of the cases, OpenSIPS as proxy (dialog stateful) is able to 
provide most of them, without the need so any fancy external servlet or 
app - just using the OpenSIPS script.


If there are good reasons to externalize the routing logic, you should 
consider more simple and flexible approach, take a look at this:

https://blog.opensips.org/2023/03/22/api-driven-sip-user-agent-end-point-with-opensips-3-4/

Regards,

Bogdan-Andrei Iancu

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

On 20.04.2024 11:50, julien.royann...@orange.com wrote:


Hello everyone,

I'm reaching out to get your opinion on using openSIPS with a business 
Application Server (AS) that would handle the routing logic.


At first glance, the SEAS module seems designed for this purpose, but 
it doesn't appear to be a good fit as it seems highly coupled with a 
Sip Servlet implementation using a specific protocol only supported 
for WeSIP.


It might be better to use REST interfaces, a 302 redirect-based 
mechanism, or possibly another module.


Thank you for your insights & advice!

JR

Orange Restricted


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Re: [OpenSIPS-Users] question on core statistics.

2024-04-24 Thread Bogdan-Andrei Iancu

Which queue are you referring at?

Regards,

Bogdan-Andrei Iancu

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

On 22.04.2024 09:17, Johan De Clercq wrote:

Goed morning,

How can I then increase opensips’s internal queue size?

Best regards, Johan

Verzonden vanuit Outlook voor iOS <https://aka.ms/o0ukef>

*Van:* Bogdan-Andrei Iancu 
*Verzonden:* Thursday, April 18, 2024 4:29:41 PM
*Aan:* OpenSIPS users mailling list ; Johan 
De Clercq 

*Onderwerp:* Re: [OpenSIPS-Users] question on core statistics.
The `drop_requests` statistic is incremented when:
* the request is dropped by a pre-script callback (like B2B when there 
is no script execution for certain messages)

* the stateless `forward()` core function failed to send out something.

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
   https://www.siphub.com  <https://www.siphub.com>
On 18.04.2024 17:19, Johan De Clercq wrote:

No I don't.
what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the 
default value of net.core.rmem_max and net.core.rmem_default.


Op do 18 apr 2024 om 16:02 schreef Ben Newlin <mailto:ben.new...@genesys.com>>:


Are you calling drop() anywhere in your script?

https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13
<https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13>

Ben Newlin

*From: *Users mailto:users-boun...@lists.opensips.org>> on behalf of Johan De
Clercq mailto:jo...@democon.be>>
*Date: *Thursday, April 18, 2024 at 5:27 AM
*To: *OpenSIPS users mailling list mailto:users@lists.opensips.org>>
*Subject: *Re: [OpenSIPS-Users] question on core statistics.

* EXTERNAL EMAIL - Please use caution with links and attachments *



would it make sense to recompile with other flags ? And how do I
set them  (I don't find these of menuconfig's compile options)?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

Op do 18 apr 2024 om 11:07 schreef Johan De Clercq
mailto:jo...@democon.be>>:

would it make sense to recompile with other flags ?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_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.

Op do 18 apr 2024 om 10:32 schreef Johan De Clercq
mailto:jo...@democon.be>>:

Guys,

I have an opensips instance running with 24 worker children.

The worker load is very low.

UDP queues are on 50 megs.

when i query via the OS

cat /proc/net/udp

  sl  local_address rem_address   st tx_queue rx_queue tr
tm->when retrnsmt   uid timeout inode ref pointer drops

  590: 03231D0A:13C4 : 07 :
00:      0   0 413684019 2
880074820bc0 0

  591: 03231D0A:13C5 : 07 :
00:      0   0 413766438 2
880465e4a440 0

  592: 03231D0A:13C6 : 07 :
00:      0   0 412035865 2
8803e5a56b80 0

  934: 01231D0A:151C : 07 :
00:      0   0 26790 2 88046c054840 0

  935: 0201FFEF:151D : 07 :
00:      0   0 26787 2 88046c054bc0 0

  935: 01231D0A:151D : 07 :
00:      0   0 26791 2 88046c0544c0 0

 1972: :D92A : 07 :
00:      0   0 15506 2 88046dce5040 0

 5479: :E6DD : 07 :
00:      0   0 22811 2 880465e4ab40 0

12075: AA0914AC:00A1 : 07 :
00:      0   0 20572 2 88086d020800 0

12075: 017F:00A1 : 07 :
00:      0   0 20571 2 88086d020b80 0

13320: :857E : 07 :
00:    100   0 17515 2 8800368ac780 0

15661: :CEA3 0

[OpenSIPS-Users] OpenSIPS 3.5 release dates

2024-04-22 Thread Bogdan-Andrei Iancu
The upcoming OpenSIPS 3.5 beta release is scheduled for *9th of May*, 
with just days before the OpenSIPS Summit in Valencia.


It focuses on #IMS (IP Multimedia Subsystem), mainly on CSCF components 
- a lot of development was done in the area and still work-in-progress. 
And, as usual, it will be the star of the Summit 2024 


https://www.opensips.org/Development/Opensips-3-5-Planning
https://www.opensips.org/events/Summit-2024Valencia/

Best regards,

--
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] question on core statistics.

2024-04-19 Thread Bogdan-Andrei Iancu
Somehow I think there is a confusion - the drop_requests stat has 
nothing to do with the dropping on the socket buffers (net level).


Regards,

Bogdan-Andrei Iancu

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

On 19.04.2024 11:13, johan wrote:


Bogdan,

on the augmenting drop_requests,

drop is not used in that script.  Neither is forward. Everything is 
t_relay.  There is no b2b.


The thing is that we observed drops on udp level.

We followed the recommendations of  Ovidiu Sas's presentation of last 
year in Houston:


- increase PKG mem

- increase SHM mem

- increase workers to 24 so that the queue empties faster.

- we checked the udp queues on linux level and we saw drops there.

    => we augmented them to 50 megs (sysctl -w 
net.core.rmem_max=52428800 and sysctl -w 
net.core.rmem_default=52428800) and the drops on OS level where gone.


Also worker and memory load are max 30 %.


Hence we thought that we were okay, but still drops on opensips 
level.  Net result was that this node in the system lost all 
connection with the destination of the loadbalancer although it 
received keep alive options responses from the loadbalancer 
destination on its NIC (we could see that in a continuously running 
tcpdump).


=> hence it seems that it is opensips's receive buffer that is too 
small (as I read  the description : "Returns the number of requests 
dropped even before entering the script routing logic.", I thought 
that this pointed to the receive buffer of opensips).  All of this is 
happening on a physical machine on which two other opensips instances 
are running also. Interestingly enough the problem is only observed in 
the instance that handles registrations and invites (1600 REG/ s and 
300 INV /s).



Therefore we dived a bit deeper and came on this MAX_RECV_BUFFER_SIZE 
262144 (which is the default udp queue size setting on linux).  Could 
this be related somehow ?



Secondly, what would the recommendation be for scaling a system like 
this ?









On 18/04/2024 16:29, Bogdan-Andrei Iancu wrote:

The `drop_requests` statistic is incremented when:
* the request is dropped by a pre-script callback (like B2B when 
there is no script execution for certain messages)

* the stateless `forward()` core function failed to send out something.

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 18.04.2024 17:19, Johan De Clercq wrote:

No I don't.
what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the 
default value of net.core.rmem_max and net.core.rmem_default.


Op do 18 apr 2024 om 16:02 schreef Ben Newlin :

Are you calling drop() anywhere in your script?

https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13

Ben Newlin

*From: *Users  on behalf of
Johan De Clercq 
*Date: *Thursday, April 18, 2024 at 5:27 AM
*To: *OpenSIPS users mailling list 
*Subject: *Re: [OpenSIPS-Users] question on core statistics.

* EXTERNAL EMAIL - Please use caution with links and attachments *



would it make sense to recompile with other flags ? And how do I
set them  (I don't find these of menuconfig's compile options)?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

Op do 18 apr 2024 om 11:07 schreef Johan De Clercq
:

would it make sense to recompile with other flags ?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE
65535.

Can somebody explain also what both flags mean.

flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_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.

Op do 18 apr 2024 om 10:32 schreef Johan De Clercq
:

Guys,

I have an opensips instance running with 24 worker
children.

The worker load is very low.

UDP queues are on 50 megs.

when i query via the OS

cat /proc/net/udp

  sl  local_address rem_address   st tx_queue rx_queue
tr tm->when retrnsmt   uid timeout inode ref pointer drops

  590: 03231D0A:13C4 : 07 :
00:      0   0 413684019 2
880074820bc0 0

  591: 03231D0A:13C5 : 07 :
00:      0   0 413766438 2
880465e4a440 0

  592: 03231D0A:13C6 : 07 :
00:      0   0 412035865 2
8803e5a56b80 0

  934: 01231D0A

Re: [OpenSIPS-Users] question on core statistics.

2024-04-18 Thread Bogdan-Andrei Iancu

The `drop_requests` statistic is incremented when:
* the request is dropped by a pre-script callback (like B2B when there 
is no script execution for certain messages)

* the stateless `forward()` core function failed to send out something.

Regards,

Bogdan-Andrei Iancu

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

On 18.04.2024 17:19, Johan De Clercq wrote:

No I don't.
what I find strange is that MAX_RECV_BUFFER_SIZE 262144 is the default 
value of net.core.rmem_max and net.core.rmem_default.


Op do 18 apr 2024 om 16:02 schreef Ben Newlin :

Are you calling drop() anywhere in your script?

https://www.opensips.org/Documentation/Script-CoreFunctions-3-4#toc13

Ben Newlin

*From: *Users  on behalf of
Johan De Clercq 
*Date: *Thursday, April 18, 2024 at 5:27 AM
*To: *OpenSIPS users mailling list 
*Subject: *Re: [OpenSIPS-Users] question on core statistics.

* EXTERNAL EMAIL - Please use caution with links and attachments *



would it make sense to recompile with other flags ? And how do I
set them  (I don't find these of menuconfig's compile options)?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

Op do 18 apr 2024 om 11:07 schreef Johan De Clercq :

would it make sense to recompile with other flags ?

Currently it has MAX_RECV_BUFFER_SIZE 262144 and BUF_SIZE 65535.

Can somebody explain also what both flags mean.

flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP,
PKG_MALLOC, F_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.

Op do 18 apr 2024 om 10:32 schreef Johan De Clercq
:

Guys,

I have an opensips instance running with 24 worker children.

The worker load is very low.

UDP queues are on 50 megs.

when i query via the OS

cat /proc/net/udp

sl  local_address rem_address   st tx_queue rx_queue tr
tm->when retrnsmt  uid  timeout inode ref pointer drops

590: 03231D0A:13C4 : 07 :
00:   0        0 413684019 2
880074820bc0 0

591: 03231D0A:13C5 : 07 :
00:   0        0 413766438 2
880465e4a440 0

592: 03231D0A:13C6 : 07 :
00:   0        0 412035865 2
8803e5a56b80 0

934: 01231D0A:151C : 07 :
00:   0        0 26790 2 88046c054840 0

935: 0201FFEF:151D : 07 :
00:   0        0 26787 2 88046c054bc0 0

935: 01231D0A:151D : 07 :
00:   0        0 26791 2 88046c0544c0 0

 1972: :D92A : 07 :
00:   0        0 15506 2 88046dce5040 0

 5479: :E6DD : 07 :
00:   0        0 22811 2 880465e4ab40 0

12075: AA0914AC:00A1 : 07 :
00:   0        0 20572 2 88086d020800 0

12075: 017F:00A1 : 07 :
00:   0        0 20571 2 88086d020b80 0

13320: :857E : 07 :
00:   100        0 17515 2 8800368ac780 0

15661: :CEA3 : 07 :
00:   0        0 15505 2 8800368acb00 0

=> no drops

what worries me is that there are drop requests and they 
go up when I query via the mi interface

opensipsctl fifo get_statistics drop_requests

core:drop_requests:: 198107

opensipsctl fifo get_statistics drop_requests

core:drop_requests:: 199157

opensipsctl_reg fifo get_statistics drop_requests

core:drop_requests:: 204116

I don't see any memory issue, also the processload is low.

so 3 questions:

- what exactly is drop_request.

- do I need to worry about this

- how can I make them go lower.

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Re: [OpenSIPS-Users] Load Balancer Probing Question

2024-04-18 Thread Bogdan-Andrei Iancu
Not really. The LB module uses internal unique ids for all the LB 
destinations it manages. So the probing replies will search back the LB 
destination based on this ID -> no chance to mismatch.


Regards.

Bogdan-Andrei Iancu

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

On 16.04.2024 10:45, Callum Guy via Users wrote:
If the backend servers are both the same instance then this seems to 
be the correct behaviour?


I believe the probing is supposed to be a simple SIP response 
healthcheck which applies to the destination globally (i.e. 1.2.3.4 is 
offline), the groups are just a way of splitting up resources 
logically for load balancing purposes.


On Tue, 16 Apr 2024 at 08:31, Bogdan-Andrei Iancu 
 wrote:


Hi,

What OpenSIPS version you have? And as I understand, as
configuration, you do permanent probing to the destinations and
the disabling happens because of this probing ?

Regards,

    Bogdan-Andrei Iancu

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

On 12.04.2024 05:41, Alexander Perkins wrote:

Hi.  I have an interesting issue.  We have two OpenSIPS servers
with load balancer (with two different group IDs in the lb table)
and we have probing set correctly and we are using the
event, E_LOAD_BALANCER_STATUS, to capture changes to servers that
were probed.  But we noticed that we have the same server URI
listed in the lb table, but with two different group IDs, if one
of the OpenSIPS servers probes that URI and it does not return,
then lb disables both groups.  I'd expect it to only disable one
group.

My question is how can we tell the LB module to disable the IP,
but also look for the groupID.  For example, I have a printout of
lb_list below.

"uri": "sip:1.2.3.4:5060 <http://1.2.3.4:5060>", "id": 27,
"group": 12,"enabled": "no", "auto-reenable": "on", "Resources":
[ { "name": "vz12", "max": 600, "load": 0 } ], "attrs": "0"

AND
{ "uri": "sip:1.2.3.4:5060 <http://1.2.3.4:5060>", "id": 29,
"group": 13,"enabled": "no", "auto-reenable": "on", "Resources":
[ { "name": "vz13", "max": 600, "load": 0 } ], "attrs": "0" },

Thank you,
Alex

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Re: [OpenSIPS-Users] Launch and async

2024-04-18 Thread Bogdan-Andrei Iancu

Hi,

The only way to do a non-blocking waiting is via the async(sleep()) 
combination, but the async() statement works only in main request route 
(in route{}) and not for reply route :( .


Doing a blocking sleep() it is very very dangerous in terms of 
performance, as you will block the opensips processes with these sleep()'s.


Anyhow, from SIP perspective it is not wise to delay a reply, as this 
will trigger retransmissions from the UAC sideso, not sure if what 
you try to do is actually something right.


Regards,

Bogdan-Andrei Iancu

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

On 16.04.2024 17:25, Parthesh Bhavsar via Users wrote:
I'm facing a challenge in my OpenSIPS implementation where I need to 
perform certain operations after receiving a 200 OK response, but only 
after waiting for 10 seconds. Currently, I'm considering using the 
sleep() function, but I'm concerned about its impact on performance.


Is there a recommended approach to achieve this functionality without 
compromising OpenSIPS performance? I'd appreciate any advice or 
alternative solutions you can offer.



Regards,

*Parthesh Bhavsar | Software Engineer | VOIP*



On Tue, Apr 16, 2024 at 7:39 PM Parthesh Bhavsar 
 wrote:


Hello,
I want to use Launch and async function for function sleep() as I
need to wait for some time to do some operations but from the
route which I use in above function I am not able to use any
variable for my operation and from documentation it seems
only able to use those variable which sleep() function use so is
there any alternative solution to get variable??? or any other
function on which I can wait for a specific time without blocking
opensips??


Regards,

*Parthesh Bhavsar | Software Engineer | VOIP*


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Re: [OpenSIPS-Users] Load Balancer Probing Question

2024-04-16 Thread Bogdan-Andrei Iancu

Hi,

What OpenSIPS version you have? And as I understand, as configuration, 
you do permanent probing to the destinations and the disabling happens 
because of this probing ?


Regards,

Bogdan-Andrei Iancu

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

On 12.04.2024 05:41, Alexander Perkins wrote:
Hi.  I have an interesting issue.  We have two OpenSIPS servers with 
load balancer (with two different group IDs in the lb table) and we 
have probing set correctly and we are using the 
event, E_LOAD_BALANCER_STATUS, to capture changes to servers that were 
probed.  But we noticed that we have the same server URI listed in the 
lb table, but with two different group IDs, if one of the OpenSIPS 
servers probes that URI and it does not return, then lb disables both 
groups.  I'd expect it to only disable one group.


My question is how can we tell the LB module to disable the IP, but 
also look for the groupID.  For example, I have a printout of lb_list 
below.


"uri": "sip:1.2.3.4:5060 <http://1.2.3.4:5060>", "id": 27, "group": 
12,"enabled": "no", "auto-reenable": "on", "Resources": [ { "name": 
"vz12", "max": 600, "load": 0 } ], "attrs": "0"


AND
{ "uri": "sip:1.2.3.4:5060 <http://1.2.3.4:5060>", "id": 29, "group": 
13,"enabled": "no", "auto-reenable": "on", "Resources": [ { "name": 
"vz13", "max": 600, "load": 0 } ], "attrs": "0" },


Thank you,
Alex

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Re: [OpenSIPS-Users] $Ri Variable

2024-04-16 Thread Bogdan-Andrei Iancu

Hi,

Check this https://opensips.org/Documentation/Script-CoreVar-3-4#socket_in

Regards,

Bogdan-Andrei Iancu

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

On 11.04.2024 20:07, Alexander Perkins wrote:
Hello!  I am trying to get the IP of the interface the request was 
received on, but I think I am doing something wrong.  When I looked at 
the Core Variables, I found $Ri.  So, I added $Ri to the cfg file, but 
now I am getting:


unknown script var $Ri, maybe a 'loadmodule' statement is missing?

Which module am I missing?  Or, am I even referencing the correct 
variable?


Thank you,
Alex

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Re: [OpenSIPS-Users] another question on sip format's.

2024-04-11 Thread Bogdan-Andrei Iancu
According to the grammar, the URI params are ONLY after the domain part 
(see https://www.ietf.org/rfc/rfc3261.html page 222)


SIP-URI  =  "sip:" [ userinfo ] hostport
uri-parameters [ headers ]
SIPS-URI =  "sips:" [ userinfo ] hostport
uri-parameters [ headers ]

Regards,

Bogdan-Andrei Iancu

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

On 11.04.2024 15:23, Johan De Clercq wrote:

Can param's be added anywhere in a user part.

e.g.
sip:0123456789;gw=case106@1.2.3.4 //this is for sure valid.
sip:gw=case107;0123456789@1.2.3.4 <mailto:case107%3B0123456789@1.2.3.4>
sip:gw=case108;0123456789;gw=case108@1.2.3.4

the fist one is for sure valid, but what about the other two ?

BR,

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Re: [OpenSIPS-Users] Message buffer formatting

2024-04-11 Thread Bogdan-Andrei Iancu
It sounds like something particular to the way you do the monitoring of 
the logs.


Bogdan-Andrei Iancu

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

On 10.04.2024 19:11, Robert Dyck wrote:

I spoke too soon. When I run opensips in debug mode on the terminal the
formatting looks good. If I monitor the log facility the $mb dump is not
formatted. Perhaps this is normal?

On Tuesday, April 9, 2024 11:47:05 P.M. PDT Bogdan-Andrei Iancu wrote:

There is no formatting added, maybe the diff comes for the actual
logging. What are the 2 versions you tested ?

Bogdan-Andrei Iancu

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

On 10.04.2024 00:21, Robert Dyck wrote:

In the past I would insert xlog with $mb into my script for debugging
purposes. Now I find that the message buffer output is not being
formatted.


Instead of


Message Buffer

REGISTER sip:192.168.1.2 SIP/2.0

Via: SIP/2.0/UDP
192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport


I get


Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via:
SIP/2.0/UDP 192.168.1.4:5070;branch=z
9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012


Instead of LF I get #012

Instead of CR I get #015


What may have caused this change and how can I restore the old behaviour?


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Re: [OpenSIPS-Users] Message buffer formatting

2024-04-10 Thread Bogdan-Andrei Iancu
There is no formatting added, maybe the diff comes for the actual 
logging. What are the 2 versions you tested ?


Bogdan-Andrei Iancu

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

On 10.04.2024 00:21, Robert Dyck wrote:


In the past I would insert xlog with $mb into my script for debugging 
purposes. Now I find that the message buffer output is not being 
formatted.



Instead of


Message Buffer

REGISTER sip:192.168.1.2 SIP/2.0

Via: SIP/2.0/UDP 
192.168.1.4:5070;branch=z9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport



I get


Message Buffer#012 REGISTER sip:192.168.1.2 SIP/2.0#015#012Via: 
SIP/2.0/UDP 192.168.1.4:5070;branch=z

9hG4bKce021e1d6a292d1504d0ff89e60c9ba;rport#015#012


Instead of LF I get #012

Instead of CR I get #015


What may have caused this change and how can I restore the old behaviour?


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Re: [OpenSIPS-Users] Waiting for 200 OK

2024-04-09 Thread Bogdan-Andrei Iancu

Hi,

my 2 cents here - how comes you want to send a RE-INVITE _BEFORE_ having 
the dialog established???


Regards,

Bogdan-Andrei Iancu

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

On 09.04.2024 17:35, Parthesh Bhavsar via Users wrote:
It seems after setting T_fr_timeout parameter it was sending a CANCEL 
request to another leg but for my requirement I need to send Reinvite. 
Also for generating Reinvite I have used dlg_send_sequential() as I 
use opensips as a proxy server. Any other modules on which I look for?

Regards,
*Parthesh Bhavsar | Software Engineer | VOIP*


On Tue, Apr 9, 2024 at 7:36 PM Ben Newlin  wrote:

The timing requirement can be solved using the $T_fr_timeout
parameter [1].

For sending a re-Invite back to the UAC I believe you’d have to be
a B2BUA.

[1] - https://opensips.org/docs/modules/3.4.x/tm.html#pv_T_fr_timeout

Ben Newlin

*From: *Users  on behalf of
Parthesh Bhavsar via Users 
*Date: *Tuesday, April 9, 2024 at 5:31 AM
*To: *OpenSIPS users mailling list 
*Subject: *[OpenSIPS-Users] Waiting for 200 OK

* EXTERNAL EMAIL - Please use caution with links and attachments *



Hello,

I have a requirement where If I get 183 responses then I need to
wait for a specific period of time for 200 OK and if 200 OK is not
received in that time then I need to send Reinvte back to UA with
some modifications. I have gone through the SST module but have
not  found anything relevant to match my requirements. Can anyone
suggest some module or function on which I can go further to meet
my requirements.


Regards,

*Parthesh Bhavsar | Software Engineer | VOIP*

*<https://www.ecosmob.com/opensips-summit/>*

*Disclaimer*

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website, any views or opinions presented in this email are solely
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Re: [OpenSIPS-Users] how to debug many dialogs stuck in state 5?

2024-04-05 Thread Bogdan-Andrei Iancu

Hi,

In the same OpenSIPS instance/script.

Regards,

Bogdan-Andrei Iancu

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

On 02.04.2024 15:57, M S wrote:

Hi Bogdan,
When you say at the same time, you mean in the same script? or same 
route? or same block for example?
Also what about using dialog and modules that use b2b in underlying 
layers, for example mediaexchange?


Thank you!

On Mon, Apr 1, 2024 at 10:28 AM Bogdan-Andrei Iancu 
 wrote:


You should never use both dialog and b2b modules in the same time,
for the same calls. Trying to have OpenSIPS both Proxy and B2B is
a clear recipe for disaster.

Regards,

Bogdan-Andrei Iancu

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

On 30.03.2024 07:53, Babak Yakhchali wrote:

sorry for the late reply. After disabling modules and simplifying
script logic I found that the issue was b2b entities module being
enabled, downgrading to 3.2 solved the problem

On Wed, Mar 20, 2024 at 5:19 PM Bogdan-Andrei Iancu
 wrote:

Hi,

What OpenSIPS version are you using? and what module do you
use on top
of the dialog module?

Regards,

Bogdan-Andrei Iancu

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

On 13.03.2024 13:55, Babak Yakhchali wrote:
> Hi
> When calling and immediately cancelling, the call is ended but
> dlg_list shows the dialog stuck in state 5. Increasing log
level to 4
> shows these messages:
> Mar 13 15:15:37 : DBG:tm:timer_routine: timer
> routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26
> Mar 13 15:15:37 : DBG:tm:delete_handler: removing
0x7f1c861d0630
> Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell
0x7f1c861d0630:
> can't delete -- still reffed (-1)
> Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2
> Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]:
0x7f1c861d06e0 (28)
> Mar 13 15:15:37 : DBG:tm:delete_handler: done
> Mar 13 15:15:39 : DBG:tm:timer_routine: timer
> routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28
> Mar 13 15:15:39 : DBG:tm:delete_handler: removing
0x7f1c861d0630
> Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell
0x7f1c861d0630:
> can't delete -- still reffed (-1)
> Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2
> Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]:
0x7f1c861d06e0 (30)
> Mar 13 15:15:39 : DBG:tm:delete_handler: done
>
> How can I debug the issue? What are the possible causes of
this?
> thanks
>
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[OpenSIPS-Users] OpenSIPS Summit 2024 - Speaker's lineup

2024-04-04 Thread Bogdan-Andrei Iancu

<https://www.opensips.org/events/Summit-2024Valencia>


 OpenSIPS Summit


 May 14th - 17th, 2024

Valencia, Spain


*Speaker's lineup & Schedule
*

We bring here the list of speakers and papers 
<https://www.opensips.org/events/Summit-2024Valencia/#schedules> - great 
speakers presenting great topics to share experience and knowledge to a 
great audience. Explore here all the details...


<https://www.opensips.org/events/Summit-2024Valencia/#schedules>

*Attend to learn* - the registration process is ongoing, the training 
class is almost full, so hurry up. The/*Corporate Package*/ is available 
with an attractive discount.


Register now <https://www.opensips.org/events/Summit-2024Valencia/#pricing>

**



    

    
    


--
Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] memory fragmentation when calling dr_reload.

2024-04-04 Thread Bogdan-Andrei Iancu

That's EOL for quite some time :(

Either consider more pkg mem (to cope with fragmentation) , either an 
upgrade to 3.4


Regards,

Bogdan-Andrei Iancu

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

On 03.04.2024 17:34, Johan De Clercq wrote:

A very old one 2.2.7

Op wo 3 apr 2024 om 16:26 schreef Bogdan-Andrei Iancu 
:


What OpenSIPS version is there ?

Regards,

Bogdan-Andrei Iancu

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

On 03.04.2024 17:04, Johan De Clercq wrote:

In addtion, I have 24 children, so can I increase in some way
only the process that handles the fifo requests ?

Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq :

Hi,

A client has a very big dynamic routing rule set. (dr_rules
>= 2.1 gb ).
When reloading the db in opensips (dr_reload), I see below
error in the log

ERROR:core:fm_malloc: not enough free pkg memory (268008864
bytes left), please increase the "-M" command line parameter!
the -M parameter is now at 256. Should I increase this to
3000 to work aroun this issue ?



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Re: [OpenSIPS-Users] dashboard stats from opensips

2024-04-03 Thread Bogdan-Andrei Iancu

Hi Jason,

Have you checked the Dashboard in OpenSIPS Control Panel ?

Regards,

Bogdan-Andrei Iancu

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

On 03.04.2024 13:54, nz deals wrote:

Hi everyone,

I'm seeking guidance on creating a dashboard. I'm considering saving 
dialog events in Redis (straight from OpenSIPS), allowing my dashboard 
to directly access the Redis cache. Do you think this is a wise 
strategy, or do you have any alternative suggestions? Any expert's 
suggestion will be highly appreciated. In fact if someone has any 
example to check, raise dialog events like call, ringing, 183, 
answered , cancel and bye etc...


Thank you

Regards,
Jason

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Re: [OpenSIPS-Users] memory fragmentation when calling dr_reload.

2024-04-03 Thread Bogdan-Andrei Iancu

What OpenSIPS version is there ?

Regards,

Bogdan-Andrei Iancu

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

On 03.04.2024 17:04, Johan De Clercq wrote:
In addtion, I have 24 children, so can I increase in some way only the 
process that handles the fifo requests ?


Op wo 3 apr 2024 om 15:33 schreef Johan De Clercq :

Hi,

A client has a very big dynamic routing rule set. (dr_rules >= 2.1
gb ).
When reloading the db in opensips (dr_reload), I see below error
in the log

ERROR:core:fm_malloc: not enough free pkg memory (268008864 bytes
left), please increase the "-M" command line parameter!
the -M parameter is now at 256. Should I increase this to 3000 to
work aroun this issue ?



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Re: [OpenSIPS-Users] Why does tracer uri=file show in/out sip messages and uri=sip only shows in?

2024-04-01 Thread Bogdan-Andrei Iancu

Hi Matthew,

If I understand correctly, if you use EITHER sip, EITHER file tracing 
(but only one at the time), it works ok. But using both in the same time 
doesn't ?


Regards,

Bogdan-Andrei Iancu

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

On 29.03.2024 19:57, Matthew Schumacher wrote:

On 3/29/24 9:08 AM, Matthew Schumacher wrote:

Hello All,

I have this config:

modparam("tracer","trace_id","[siptrace]uri=sip:127.0.0.1:")
modparam("tracer","trace_id","[siptrace]uri=file:/var/log/siptrace.log")

route {
    trace("siptrace","d","sip");

...

When I look at the messages that show up on UDP: I only see 
egress sip messages, but when I look at /var/log/siptrace.log I see 
both sides of the conversation.  Is this by design?  Is there a way 
to send all messages to a port, or a socket, or perhaps define a pipe 
to another process?  I'm trying to setup a second daemon that 
monitors all of the sip dialog between opensips and other hosts.


Thanks!
Matt




Correction, I only see ingress SIP messages, not egress when using the 
sip: uri in the tracer module.  If I use file: or hep: I see both sides.




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Re: [OpenSIPS-Users] How to access to a column of a location database

2024-04-01 Thread Bogdan-Andrei Iancu

Hi,

With that approach you still have to do the actual lookup(), something 
that Guillaume wants to avoid, AFAIU. I mean if you do end up doing 
lookup(), you can check the $socket_out directly :)


If you want to fetch the socket info WITHOUT doing lookup() (and 
affecting the current SIP message), you can do this via the MI 
`ul_show_contct` [1] directly via script using the mi_script module [2]



[1] 
https://opensips.org/html/docs/modules/3.4.x/usrloc.html#mi_ul_show_contact

[2] https://opensips.org/html/docs/modules/3.4.x/mi_script.html#func_mi

Regards,

Bogdan-Andrei Iancu

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

On 29.03.2024 16:01, Vlad Paiu wrote:


Hello,

You can use 
https://opensips.org/html/docs/modules/3.3.x/registrar.html#param_attr_avp 
in order to populate any custom info at save() time and the info will 
automatically be populated for you at lookup() time.



Regards,
Vlad


On 29.03.2024 15:08, guillaume.desgeo...@orange.com wrote:


Hi Bogdan,

Thank you for your answer.

And is it possible to use that lookup("location") function to put the 
“registered socket” field in a variable in order to use it for my 
script ?


Regards,

  Guillaume

*De :*Bogdan-Andrei Iancu 
*Envoyé :* jeudi 28 mars 2024 11:15
*À :* OpenSIPS users mailling list ; 
DESGEORGE Guillaume INNOV/IT-S 
*Objet :* Re: [OpenSIPS-Users] How to access to a column of a 
location database


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Hi Guillaume,

The registered contact (and the additional info) is fetched via the 
lookup("location") function. It also sets the registered socket for 
the routing the current request.


Regards,

Bogdan-Andrei Iancu
  
OpenSIPS Founder and Developer

   https://www.opensips-solutions.com
   https://www.siphub.com

On 22.03.2024 10:57, guillaume.desgeo...@orange.com wrote:

Hi everybody,

In my routing logic of my “opensips.cfg” file, I’m trying to
access the “socket” field of the location table of registered
contacts.

I have the registering informations correctly written in MySQL
location table but can’t find a function to access it.

I’d like to have a function which I give the registered username
and can give me back the associated socket.

Is the lookup () function the good one ? I didn’t understand how
to use it that way.

Thanks for your help,

   Guillaume

Orange Restricted

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Re: [OpenSIPS-Users] Opensips 3.2 with TCP connintion and DB operation on a single instance .

2024-04-01 Thread Bogdan-Andrei Iancu

Hi,

If you are referring to the "TCP Connect Issues", yes, it is still valid.

Regards,

Bogdan-Andrei Iancu

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

On 29.03.2024 14:07, Sasmita Panda wrote:

Hi All ,

Earlier in the very early stage of using opensips I had faced some 
issues while using TCP with DB operation . Opensips has evolved a lot 
in the past few years with so many additional features .


Now , can I use TCP globally with opensips 3.2 . My config will do DB 
lookup as well .


https://www.opensips.org/Documentation/Script-Async-3-2#toc10
Is this relevant to my scenario or what issue is mentioned here  ?

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/Sasmita Panda/
/Senior Network Testing and Software Engineer/
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Re: [OpenSIPS-Users] REINVITE IN OPENSIPS

2024-04-01 Thread Bogdan-Andrei Iancu

Hi,

Check this dlg_send_requential() [1] function.

[1] 
https://opensips.org/html/docs/modules/3.4.x/dialog.html#func_dlg_send_sequential


Regards,

Bogdan-Andrei Iancu

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

On 01.04.2024 09:06, Parthesh Bhavsar via Users wrote:

Hi All,
I have a requirement where I need to send ReInvite from opensips after 
receiving 200 OK. So First I need to confirm whether Opensips is able 
to send ReInvite ?? I have gone through module b2b_logic and 
b2b_entities for the same and it seems opensips sends ReInvite for 
late SDP negotiations but I want to send after successful bridge for 
some other task.


Can anyone suggest to me which module I need to spend my time on to 
meet my requirements ??


Regards,

*Parthesh Bhavsar | Software Engineer | VOIP*
*+91 9638867145*

*Ecosmob Technologies Pvt Ltd.* <https://www.ecosmob.com/>

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Re: [OpenSIPS-Users] how to debug many dialogs stuck in state 5?

2024-04-01 Thread Bogdan-Andrei Iancu
You should never use both dialog and b2b modules in the same time, for 
the same calls. Trying to have OpenSIPS both Proxy and B2B is a clear 
recipe for disaster.


Regards,

Bogdan-Andrei Iancu

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

On 30.03.2024 07:53, Babak Yakhchali wrote:
sorry for the late reply. After disabling modules and simplifying 
script logic I found that the issue was b2b entities module being 
enabled, downgrading to 3.2 solved the problem


On Wed, Mar 20, 2024 at 5:19 PM Bogdan-Andrei Iancu 
 wrote:


Hi,

What OpenSIPS version are you using? and what module do you use on
top
of the dialog module?

Regards,

Bogdan-Andrei Iancu

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

On 13.03.2024 13:55, Babak Yakhchali wrote:
> Hi
> When calling and immediately cancelling, the call is ended but
> dlg_list shows the dialog stuck in state 5. Increasing log level
to 4
> shows these messages:
> Mar 13 15:15:37 : DBG:tm:timer_routine: timer
> routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26
> Mar 13 15:15:37 : DBG:tm:delete_handler: removing 0x7f1c861d0630
> Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630:
> can't delete -- still reffed (-1)
> Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2
> Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]:
0x7f1c861d06e0 (28)
> Mar 13 15:15:37 : DBG:tm:delete_handler: done
> Mar 13 15:15:39 : DBG:tm:timer_routine: timer
> routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28
> Mar 13 15:15:39 : DBG:tm:delete_handler: removing 0x7f1c861d0630
> Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630:
> can't delete -- still reffed (-1)
> Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2
> Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]:
0x7f1c861d06e0 (30)
> Mar 13 15:15:39 : DBG:tm:delete_handler: done
>
> How can I debug the issue? What are the possible causes of this?
> thanks
>
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Re: [OpenSIPS-Users] Callee details not visible in dialog detail

2024-03-28 Thread Bogdan-Andrei Iancu

Hi,

If using Kamailio, you should ask this on the kamilio mailing list, 
right ? This mailing list is for OpenSIPS.


Regards

Bogdan-Andrei Iancu

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

On 28.03.2024 16:35, egemen ulus wrote:

Hi,
I am wondering why callee info is not visible in dialog details:
Here is the case:

client-1 > kamailio > two asterisks > kamailio > client-2
registration is on asterisk.

dialog module is enabled and I want to see the asterisk IP that is 
determined by dispatcher module.
I have tried to put dlg_manage function even after dispatch function, 
but did not help.


Any way to have asterisk IP in dialog details?

It shows only this:

{
 h_entry: 483
 h_id: 8644
 ref: 1
 call-id: v489LdfOFD5mnlHupQHOvw..
 from_uri: sip:1515@12.12.12.12;transport=UDP
 to_uri: sip:@12.12.12.12
 state: 5
 start_ts: 0
 init_ts: 
 end_ts: 
 duration: 
 timeout: 0
 lifetime: 3600
 dflags: 512
 sflags: 0
 iflags: 0
 caller: {
  tag: d0283007
  contact: sip:1515@1.1.1.1:64966;transport=UDP
  cseq: 1
  route_set:
  socket: udp:172.22.10.10:5060
 }
 callee: {
  tag: 
  contact: 
  cseq: 
  route_set: 
  socket: 
 }
 profiles: {
 }
 variables: {
 }
}


Regards

Get Outlook for Android <https://aka.ms/AAb9ysg>

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Re: [OpenSIPS-Users] https://datatracker.ietf.org/doc/html/rfc3261#section-13.3.1.1

2024-03-28 Thread Bogdan-Andrei Iancu

For any reply, see here the condition:

https://github.com/OpenSIPS/opensips/blob/4b23a80bd14dcf509ebe8de22f26906d34e0b079/modules/tm/t_reply.c#L1697

Keep in mind that the restart_fr_on_each_reply is by default on (set to 1).


Regards,

Bogdan-Andrei Iancu

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

On 28.03.2024 12:37, Johan De Clercq wrote:

Thanks for the swift reply Bogdan.
but shouldn't this timer be reset when receiving f.e. 180 or 183 ?


Op do 28 mrt 2024 om 11:35 schreef Bogdan-Andrei Iancu 
:


Hi Johan,

In OpenSIPS, the fr_inv_timer kicks in (for INVITEs) when a first
reply (typically a 100) is received from UAS. And it will wait for
the final reply. So, it is how long to wait from the first reply
up to the final one.

Regards,

Bogdan-Andrei Iancu

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

On 28.03.2024 10:29, Johan De Clercq wrote:

Question,

I always believed that fr_inv_timer should trigger when an invite
is not finished in due time.
This seems however to contradict the link in the title

  If the UAS is not able to answer the invitation immediately, it can
choose to indicate some kind of progress to the UAC (for example, an
indication that a phone is ringing).  This is accomplished with a
provisional response between 101 and 199.  These provisional
responses establish early dialogs and therefore follow the procedures
ofSection 12.1.1  
<https://datatracker.ietf.org/doc/html/rfc3261#section-12.1.1>  in addition to those 
ofSection 8.2.6  <https://datatracker.ietf.org/doc/html/rfc3261#section-8.2.6>.  A 
UAS MAY
send as many provisional responses as it likes.  Each of these MUST
indicate the same dialog ID.  However, these will not be delivered
reliably.

If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction.  A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction.  To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.

   An INVITE transaction can go on for extended durations when the
   user is placed on hold, or when interworking with PSTN systems
   which allow communications to take place without answering the
   call.  The latter is common in Interactive Voice Response (IVR)
   systems.

Can somebody please comment on this ?

BR, Johan.


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Re: [OpenSIPS-Users] https://datatracker.ietf.org/doc/html/rfc3261#section-13.3.1.1

2024-03-28 Thread Bogdan-Andrei Iancu

Hi Johan,

In OpenSIPS, the fr_inv_timer kicks in (for INVITEs) when a first reply 
(typically a 100) is received from UAS. And it will wait for the final 
reply. So, it is how long to wait from the first reply up to the final one.


Regards,

Bogdan-Andrei Iancu

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

On 28.03.2024 10:29, Johan De Clercq wrote:

Question,

I always believed that fr_inv_timer should trigger when an invite is 
not finished in due time.

This seems however to contradict the link in the title

  If the UAS is not able to answer the invitation immediately, it can
choose to indicate some kind of progress to the UAC (for example, an
indication that a phone is ringing).  This is accomplished with a
provisional response between 101 and 199.  These provisional
responses establish early dialogs and therefore follow the procedures
ofSection 12.1.1  <https://datatracker.ietf.org/doc/html/rfc3261#section-12.1.1>  
in addition to those ofSection 8.2.6  
<https://datatracker.ietf.org/doc/html/rfc3261#section-8.2.6>.  A UAS MAY
send as many provisional responses as it likes.  Each of these MUST
indicate the same dialog ID.  However, these will not be delivered
reliably.

If the UAS desires an extended period of time to answer the INVITE,
it will need to ask for an "extension" in order to prevent proxies
from canceling the transaction.  A proxy has the option of canceling
a transaction when there is a gap of 3 minutes between responses in a
transaction.  To prevent cancellation, the UAS MUST send a non-100
provisional response at every minute, to handle the possibility of
lost provisional responses.

   An INVITE transaction can go on for extended durations when the
   user is placed on hold, or when interworking with PSTN systems
   which allow communications to take place without answering the
   call.  The latter is common in Interactive Voice Response (IVR)
   systems.

Can somebody please comment on this ?

BR, Johan.


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Re: [OpenSIPS-Users] Getting the $rb or media port for egress message.

2024-03-28 Thread Bogdan-Andrei Iancu

Hi Matthew,

For efficiency reasons, the changes over the SIP messages are not 
applied (so vizible) on the spot, but only at the end, when the SIP 
message is sent out. So, shortly, you cannot see the changes done by 
your script, you all the time "see" the initial received message.


Options you have here:

1) force applying the changes, see the get_updated_body_part() [1]
  if (has_body("application/sdp")){
    rtpengine_offer();
    get_updated_body_part( "application/sdp", $var(new_sdp));
    $var(mline) = $(var(new_sdp){sdp.line,m,0});
    xlog("TEST $var(mline)\n");
  }

2) instruct rtpengine not to change the SDP in the message, but to do 
the changes over an SDP stored in a variable. And you will push the new 
SDP manually in the message.

  if (has_body("application/sdp")){
$var(new_sdp)= $rb;
    rtpengine_offer(,$var(new_sdp));
    $var(mline) = $(var(new_sdp){sdp.line,m,0});
    remove_body_part("application/sdp");
    add_body_part($var(new_sdp),"application/sdp");
    xlog("TEST $var(mline)\n");
  }


[1] 
https://opensips.org/html/docs/modules/3.4.x/sipmsgops.html#func_get_updated_body_part


Regards,

Bogdan-Andrei Iancu

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

On 26.03.2024 23:17, Matthew Schumacher wrote:

Hello,

I'm trying to log the media port for another system and using this 
code to grab the m line out of an SDP header:


  if (has_body("application/sdp")){
    rtpengine_offer();
    $var(mline) = $(rb{sdp.line,m,0});
    xlog("TEST $var(mline)\n");
  }

The problem is that it only sees the ports for ingress SDP messages 
and not egress SDP messages.  What can I do to grab the SDP messages 
leaving opensips?


I want to take this data and inject a firewall rule that allows the 
media and remove it later because NAT helpers don't work on TLS 
traffic, my system already knows what port to expect, and I don't like 
having a huge pile of UDP ports allowed through.


Honestly, I'm not sure why the kernel mode forwarding in rtpengine 
doesn't also allow traffic based on ports we are expecting to see 
traffic from that's what I thought it did at first until I 
realized that it's only forwarding the RTP packet through the kernel 
faster without needing userspace.


Thanks,
Matt

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Re: [OpenSIPS-Users] How to access to a column of a location database

2024-03-28 Thread Bogdan-Andrei Iancu

Hi Guillaume,

The registered contact (and the additional info) is fetched via the 
lookup("location") function. It also sets the registered socket for the 
routing the current request.


Regards,

Bogdan-Andrei Iancu

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

On 22.03.2024 10:57, guillaume.desgeo...@orange.com wrote:


Hi everybody,

In my routing logic of my “opensips.cfg” file, I’m trying to access 
the “socket” field of the location table of registered contacts.


I have the registering informations correctly written in MySQL 
location table but can’t find a function to access it.


I’d like to have a function which I give the registered username and 
can give me back the associated socket.


Is the lookup () function the good one ? I didn’t understand how to 
use it that way.


Thanks for your help,

   Guillaume

Orange Restricted


Orange Restricted


Ce message et ses pieces jointes peuvent contenir des informations 
confidentielles ou privilegiees et ne doivent donc
pas etre diffuses, exploites ou copies sans autorisation. Si vous avez recu ce 
message par erreur, veuillez le signaler
a l'expediteur et le detruire ainsi que les pieces jointes. Les messages 
electroniques etant susceptibles d'alteration,
Orange decline toute responsabilite si ce message a ete altere, deforme ou 
falsifie. Merci.

This message and its attachments may contain confidential or privileged 
information that may be protected by law;
they should not be distributed, used or copied without authorisation.
If you have received this email in error, please notify the sender and delete 
this message and its attachments.
As emails may be altered, Orange is not liable for messages that have been 
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[OpenSIPS-Users] If it's May, it's OpenSIPS Summit

2024-03-26 Thread Bogdan-Andrei Iancu


Spring is here and May is getting closer and closer - and we celebrate 
this OpenSIPS Summit vibe with the *summit-24-flash-sale* 50% discount 
code for all registrations between 28th 00:00 and 31th 23:59 GMT - we 
love March . This special offer is limited to 20 seats, so hurry up !


https://www.opensips.org/events/Summit-2024Valencia/#pricing

See you in Valencia!

--
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-03-22 Thread Bogdan-Andrei Iancu

IF

(a) it crashes, try to get a backtrace

(b) it block on starting, try to do a "trap" via opensips-cli

Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:24, Sasmita Panda wrote:
Sometimes it crashes and sometimes while starting I get the warings of 
the timer .  Same config  shows different issues .




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


On Wed, Mar 20, 2024 at 6:45 PM Bogdan-Andrei Iancu 
 wrote:


Hi,

How the two reports fit together here ? there are completely
separate experiences on different runs?? or if you start opensips
first you get the warnings and later it crashes ??
For the crash part, I see a core file was generated - could you
extract the backtrace and post here ? (see
https://opensips.org/Documentation/TroubleShooting-Crash)

    Regards

    Bogdan-Andrei Iancu

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

On 11.03.2024 15:04, Sasmita Panda wrote:

Any update on this ?


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


On Mon, Mar 11, 2024 at 12:03 PM Sasmita Panda
 wrote:

With the same server configuration and opensips version I am
getting below error as well .

CRITICAL:core:fm_free: freeing already freed shm pointer
(0x7fc110e0b408), first free: (null): (null)(0) - aborting!

 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs:
child process 215185 exited by a signal 6
 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs:
core was generated
 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs:
terminating due to SIGCHLD
 /usr/local/sbin/opensips[215172]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: fm_status (0x7fc2907ff010):
 /usr/local/sbin/opensips[215174]: INFO:core:sig_usr: signal
15 received
 /usr/local/sbin/opensips[215172]: heap size= 33554432
 /usr/local/sbin/opensips[215174]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: used= 3710048,
used+overhead=3801344, free=29844384
 /usr/local/sbin/opensips[215174]: fm_status (0x7fc2907ff010):
 /usr/local/sbin/opensips[215172]: max used (+overhead)= 3801344
 /usr/local/sbin/opensips[215174]: heap size= 33554432
 /usr/local/sbin/opensips[215172]: dumping free list:
 /usr/local/sbin/opensips[215174]: used= 229752,
used+overhead=318936, free=33324680
 /usr/local/sbin/opensips[215172]: hash =   1 fragments no.:
    1, unused:     0#012#011#011 bucket size:         8 -    
    8 (first 8)
 /usr/local/sbin/opensips[215174]: max used (+overhead)= 385792
 /usr/local/sbin/opensips[215174]: dumping free list:
 /usr/local/sbin/opensips[215172]: hash =  18 fragments no.:
    1, unused:     0#012#011#011 bucket size:       144 -    
  144 (first 144)
 /usr/local/sbin/opensips[215174]: hash =   7 fragments no.:
  139, unused:     0#012#011#011 bucket size:        56 -    
   56 (first  56)
 /usr/local/sbin/opensips[215172]: hash = 2059 fragments no.:
    1, unused:     0#012#011#011 bucket size:  16777216 -
 33554432 (first  29752936)
 /usr/local/sbin/opensips[215174]: hash =  13 fragments no.:
   37, unused:     0#012#011#011 bucket size:       104 -    
  104 (first 104)
 /usr/local/sbin/opensips[215172]: TOTAL:      3 free
fragments = 29753088 free bytes
 /usr/local/sbin/opensips[215172]: TOTAL: 48 overhead
 /usr/local/sbin/opensips[215174]: hash =  16 fragments no.:
   61, unused:     0#012#011#011 bucket size:       128 -    
  128 (first 128)
 /usr/local/sbin/opensips[215172]: -
 /usr/local/sbin/opensips[215174]: hash =  31 fragments no.:
  152, unused:     0#012#011#011 bucket size:       248 -    
  248 (first 248)
 /usr/local/sbin/opensips[215174]: hash =  68 fragments no.:
   16, unused:     0#012#011#011 bucket size:       544 -    
  544 (first 544)
 /usr/local/sbin/opensips[215174]: hash = 105 fragments no.:
    1, unused:     0#012#011#011 bucket size:       840 -    
  840 (first 840)
 /usr/local/sbin/opensips[215174]: hash = 2059 fragments no.:
    1, unused:     0#012#011#011 bucket size:  16777216 -
 33554432 (first  33168816)
 /usr/local/sbin/opensips[215174]: TOTAL:    407 free
fragments = 33235496 free bytes
 /usr/local/sbin/opensips[215174]: TOTAL: 48 overhead
 /usr/local/sb

Re: [OpenSIPS-Users] SDP

2024-03-22 Thread Bogdan-Andrei Iancu
On 2.2 you cannot use variables as parms for any script functions, some 
of them do not support variables. And replace_body() is one of them.


In 3.4 all script functions can take vars as params, so you should be fine.

Regards,

Bogdan-Andrei Iancu

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

On 22.03.2024 08:37, Stefan Carlsson wrote:
Still on 2.2 but are planning to migrste later in the spring to Ubuntu 
and 3.4   ..


Can you please in some quick notes tell me what must change in the 
script if required

Regards …

Stefan


*From:* Bogdan-Andrei Iancu 
*Sent:* Thursday, 21 March 2024 17:30
*To:* Stefan Carlsson; users@lists.opensips.org
*Subject:* Re: [OpenSIPS-Users] SDP

Hi,

What OpenSIPS version are you using?

Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 21.03.2024 00:09, Stefan Carlsson wrote:


Hi !

I sincerely apologize that I not read thru all the modules before my 
orginal post , I’ve found that textops module can search and replace


the text in both the header and the body. (which I mailed to the 
requester)


I’ve tried some of the exported functions but:

orginal snap:  o=root 276479022 276479022 IN IP4 172.24.7.20

$avp(reinv_sess_id) = “12345678”;    # Got this from  
$avp(reinv_oline) = $(rb{sdp.line,o})   and some regex tweeking and 
 $(avp(reinv_result){s.select,1,;})


$avp(reinv_sess_ver) = “276479022”;

replace_body("$avp(reinv_sess_ver)","$avp(reinv_sess_id)");

This seemed not to work …

but this did:

Original text snap:    a=sendrecv

replace_body("a=sendrecv", "a=sendonly");   # Indicate call hold 
according RFC3264


// Regards …

Stefan

*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, 20 March, 2024 14:44
*To:* OpenSIPS users mailling list ; Stefan 
Carlsson 

*Subject:* Re: [OpenSIPS-Users] SDP

Nothing is impossible as time as you search in the right place (aka 
the Manual). See my reply on your thread.


Regards,

Bogdan-Andrei Iancu
  
OpenSIPS Founder and Developer

   https://www.opensips-solutions.com
   https://www.siphub.com

On 14.03.2024 09:50, Stefan Carlsson wrote:

Hi !

You can use the module sipmsgops to view and change some of the
ADP fields such as codecs and streams … but unfortunately you
don’t have

complete control of the SDP compared to what it seems could be
done on the Kamailio SIP router

I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users 
<mailto:users-boun...@lists.opensips.org> *On Behalf Of *Prathibha B
*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

-- 


Regards,

B.Prathibha



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Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed

2024-03-21 Thread Bogdan-Andrei Iancu

HI Alan,

I suspect a scripting issue, making the received hop-by-hop ACK (for 
negative replies) to be "handled" by the dialog module, which treats it 
as a end-2-end ACK (for 200 OK reply).


I guess your script lacks this block 
https://github.com/OpenSIPS/opensips/blob/master/etc/opensips.cfg#L112 
when comes to sequential requests; this block needs to be before the 
loose_route or any dialog/TH matching.


Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 12:29, Alain Bieuzent wrote:


Hi bogdan,

Find attached the sip traces where we have this error log:

Mar 21 11:24:13 lbsip-rtpe-test opensips[3108]: 
WARNING:dialog:dlg_onroute: tight matching failed for ACK with 
callid='6a543fbd6eabdc7b797dfd6d56db1568@10.101.180.176:5060'/52, 
ftag='as5625de52'/10, ttag='b5a7-94b8b0d1dd76318f3d174a911be4ece2'/37 
and direction=1


Mar 21 11:24:13 lbsip-rtpe-test opensips[3108]: 
WARNING:dialog:dlg_onroute: dialog identification elements are 
callid='6a543fbd6eabdc7b797dfd6d56db1568@10.101.180.176:5060'/52, 
caller tag='as5625de52'/10, callee tag='3920005391-2017127892'/21


*De : *Bogdan-Andrei Iancu 
*Date : *mercredi 20 mars 2024 à 16:45
*À : *Alain Bieuzent , OpenSIPS users mailling 
list 
*Objet : *Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight 
matching failed


Not the logs, but the trace/ SIP capture - I need to see the sip 
traffic to understand the context of the issue.



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

On 20.03.2024 17:14, Alain Bieuzent wrote:

Bellow the link to the traces :

https://www.transfernow.net/dl/20240320jzEA98Vz

    *De : *Bogdan-Andrei Iancu 
<mailto:bog...@opensips.org>
*Date : *mercredi 20 mars 2024 à 14:29
*À : *OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>, Alain Bieuzent
 <mailto:alain.bieuz...@free.fr>
*Objet : *Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight
matching failed

Hi,

The script snippet has nothing to do with the matching - it is
about a 200-ok ACK to be matched by the dialog module and the
TO-tag does not match at all.

The link to the trace expired, if you could repost.

Regards,


Bogdan-Andrei Iancu

  


OpenSIPS Founder and Developer

   https://www.opensips-solutions.com

   https://www.siphub.com

On 12.03.2024 14:57, Alain Bieuzent wrote:

Hi all,

I am in the process of migrating my proxies from 3.3.9 to 3.4.4.

In 3.4 I find myself with a fault when I try to rewrite a
final cause.

if (t_check_status("500"))

 {

send_reply(503,"Service Unavailable");

exit;

}

with 3.3 version, this code works perfectly, since 3.4 I have
this error:

WARNING:dialog:dlg_onroute: tight matching failed for ACK with
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52,
ftag='as5328cea7'/10, ttag='b5a7-2a346bb9d510da893a
ee7cda850584cb'/37 and direction=1

WARNING:dialog:dlg_onroute: dialog identification elements are
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52,
caller tag='as5328cea7'/10, callee tag='3919223738-770090972'/ 20

Has anyone encountered this error before?

Complete traces here :
https://www.transfernow.net/dl/20240312qvAFfIAL.

Regards




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Re: [OpenSIPS-Users] Need some help on opensips error for wss handshek .

2024-03-21 Thread Bogdan-Andrei Iancu

Just try the pointed patch, to see if it solves your problem..

Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 08:37, Sasmita Panda wrote:
We are using webrtc Client integrated with browsers using sipml5 
client library . I think this is a common library used across the glob .
Every customer that has this same client running with webrtc Server 
(having opensips) is common .

But the error I have seen is for one or two agents only .


So the error is for the client or the network ? If it can be the 
network then I may suggest the customer to change the network while 
using webrtc client .



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


On Wed, Mar 20, 2024 at 7:55 PM Bogdan-Andrei Iancu 
 wrote:


Hi,

There is a fix (to tolerate different version numbers there) but
only in OpenSIPS 3.5 [1] (the devel version), not yet backported
to stable 3.4 and 3.2. You can try to do the backport by yourself.

BTW, what WS client & browser are you using ?? It is pretty
unusual what you report here.

[1]

https://github.com/OpenSIPS/opensips/commit/e30a355397b285a5140ebb25a485bed7134de7ee

    Bogdan-Andrei Iancu

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

On 20.03.2024 15:30, Sasmita Panda wrote:

Hi Bogdan,
I appreciate your reply .

How can I suggest modifying this header ? Because I have seen
this for some remote agents who run in the browser . I don't have
access to the browser .
The browser which got connected successfully I couldn't see such
types of logs  .

What causes this header version in the wss connection ?


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


On Wed, Mar 20, 2024 at 6:51 PM Bogdan-Andrei Iancu
 wrote:

Hi,

OpenSIPS does not like that 1.1 version in the VIA hdr - it
expects 2.0 (the SIP version).

Regards,

    Bogdan-Andrei Iancu

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

On 12.03.2024 10:57, Sasmita Panda wrote:

Hi All ,

1"  :callID 080a9ef8-3911-6c41-689c-fbb5d02e3aae
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via: bad char <1> on state 100
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via:  <1.1 p1wsa01.rivagecapital.com:80
<http://p1wsa01.rivagecapital.com:80>
(Cisco-WSA/12.5.4-011)#015#012#015#012>
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via: via parse failed
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:get_hdr_field: bad via
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
INFO:core:parse_headers: bad header field
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:proto_wss:ws_parse_req_handshake: cannot parse
headers#012Upgrade: websocket#015#012Connection:
Upgrade#015#012Host: ir-webrtc2.3ccloud.com#015#012Pragma
<http://ir-webrtc2.3ccloud.com#015%23012Pragma>:
no-cache#015#012Cache-Control: no-cache#015#012User-Agent:
Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36
(KHTML, like Gecko) Chrome/122.0.0.0 <http://122.0.0.0>
Safari/537.36 Edg/122.0.0.0#015#012Origin
<http://122.0.0.0#015%23012Origin>:
https://ir-static.3ccloud.com#015#012Sec-WebSocket-Version
<https://ir-static.3ccloud.com#015%23012Sec-WebSocket-Version>:
13#015#012Accept-Encoding: gzip, br#015#012Accept-Language:
en-GB,en;q=0.9,en-US;q=0.8#015#012Sec-WebSocket-Key:
I/8XqtXyBCAGorZYjnTYQA==#015#012Sec-WebSocket-Extensions:
permessage-deflate;
client_max_window_bits#015#012Sec-WebSocket-Protocol:
sip#015#012X-IMForwards: 20#015#012Via: 1.1
p1wsa01.rivagecapital.com:80
<http://p1wsa01.rivagecapital.com:80>
(Cisco-WSA/12.5.4-011)#015#012#015#012


 I am getting this error on my opensips server where I am
running webrtc . What is this error for ?


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

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Re: [OpenSIPS-Users] SDP

2024-03-21 Thread Bogdan-Andrei Iancu

Hi,

What OpenSIPS version are you using?

Regards,

Bogdan-Andrei Iancu

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

On 21.03.2024 00:09, Stefan Carlsson wrote:


Hi !

I sincerely apologize that I not read thru all the modules before my 
orginal post , I’ve found that textops module can search and replace


the text in both the header and the body. (which I mailed to the 
requester)


I’ve tried some of the exported functions but:

orginal snap:  o=root 276479022 276479022 IN IP4 172.24.7.20

$avp(reinv_sess_id) = “12345678”;    # Got this from  
$avp(reinv_oline) = $(rb{sdp.line,o})   and some regex tweeking and 
 $(avp(reinv_result){s.select,1,;})


$avp(reinv_sess_ver) = “276479022”;

replace_body("$avp(reinv_sess_ver)","$avp(reinv_sess_id)");

This seemed not to work …

but this did:

Original text snap:    a=sendrecv

replace_body("a=sendrecv", "a=sendonly");   # Indicate call hold 
according RFC3264


// Regards …

Stefan

*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, 20 March, 2024 14:44
*To:* OpenSIPS users mailling list ; Stefan 
Carlsson 

*Subject:* Re: [OpenSIPS-Users] SDP

Nothing is impossible as time as you search in the right place (aka 
the Manual). See my reply on your thread.


Regards,

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

On 14.03.2024 09:50, Stefan Carlsson wrote:

Hi !

You can use the module sipmsgops to view and change some of the
ADP fields such as codecs and streams … but unfortunately you
don’t have

complete control of the SDP compared to what it seems could be
done on the Kamailio SIP router

I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users 
<mailto:users-boun...@lists.opensips.org> *On Behalf Of *Prathibha B
*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
<mailto:users@lists.opensips.org>
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

-- 


Regards,

B.Prathibha



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Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed

2024-03-20 Thread Bogdan-Andrei Iancu
Not the logs, but the trace/ SIP capture - I need to see the sip traffic 
to understand the context of the issue.


Bogdan-Andrei Iancu

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

On 20.03.2024 17:14, Alain Bieuzent wrote:


Bellow the link to the traces :

https://www.transfernow.net/dl/20240320jzEA98Vz

*De : *Bogdan-Andrei Iancu 
*Date : *mercredi 20 mars 2024 à 14:29
*À : *OpenSIPS users mailling list , Alain 
Bieuzent 
*Objet : *Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight 
matching failed


Hi,

The script snippet has nothing to do with the matching - it is about a 
200-ok ACK to be matched by the dialog module and the TO-tag does not 
match at all.


The link to the trace expired, if you could repost.

Regards,

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

On 12.03.2024 14:57, Alain Bieuzent wrote:

Hi all,

I am in the process of migrating my proxies from 3.3.9 to 3.4.4.

In 3.4 I find myself with a fault when I try to rewrite a final cause.

if (t_check_status("500"))

 {

send_reply(503,"Service Unavailable");

exit;

}

with 3.3 version, this code works perfectly, since 3.4 I have this
error:

WARNING:dialog:dlg_onroute: tight matching failed for ACK with
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52,
ftag='as5328cea7'/10, ttag='b5a7-2a346bb9d510da893a
ee7cda850584cb'/37 and direction=1

WARNING:dialog:dlg_onroute: dialog identification elements are
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52,
caller tag='as5328cea7'/10, callee tag='3919223738-770090972'/ 20

Has anyone encountered this error before?

Complete traces here :
https://www.transfernow.net/dl/20240312qvAFfIAL.

Regards



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Re: [OpenSIPS-Users] call center

2024-03-20 Thread Bogdan-Andrei Iancu
Right, see the other thread, with the external app. And for the future 
try to have only one email thread per topic, do not flood with multiple 
emails on the same topic.


Bogdan-Andrei Iancu

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

On 20.03.2024 16:29, Prathibha B wrote:
 I want to pass the call START, CONNECT, END messages to the external 
application


Sent from Outlook for Android <https://aka.ms/AAb9ysg>

*From:* Prathibha B 
*Sent:* Wednesday, March 20, 2024 7:56:14 PM
*To:* Bogdan-Andrei Iancu ; OpenSIPS users 
mailling list 

*Subject:* Re: [OpenSIPS-Users] call center
Need to pass data from opensips to external call center application 
like call ID, duration of the call and so on.


Sent from Outlook for Android <https://aka.ms/AAb9ysg>
----
*From:* Bogdan-Andrei Iancu 
*Sent:* Wednesday, March 20, 2024 7:51:06 PM
*To:* OpenSIPS users mailling list ; 
Prathibha B 

*Subject:* Re: [OpenSIPS-Users] call center
Could you be more specific on your question, like what OpenSIPS should 
do for the call center application??


Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com  <https://www.opensips-solutions.com>
   https://www.siphub.com  <https://www.siphub.com>
On 19.03.2024 07:03, Prathibha B wrote:

How to integrate Opensips with call center application?

--
Regards,
B.Prathibha

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Re: [OpenSIPS-Users] Need some help on opensips error for wss handshek .

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

There is a fix (to tolerate different version numbers there) but only in 
OpenSIPS 3.5 [1] (the devel version), not yet backported to stable 3.4 
and 3.2. You can try to do the backport by yourself.


BTW, what WS client & browser are you using ?? It is pretty unusual what 
you report here.


[1] 
https://github.com/OpenSIPS/opensips/commit/e30a355397b285a5140ebb25a485bed7134de7ee


Bogdan-Andrei Iancu

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

On 20.03.2024 15:30, Sasmita Panda wrote:

Hi Bogdan,
I appreciate your reply .

How can I suggest modifying this header ? Because I have seen this for 
some remote agents who run in the browser . I don't have access to the 
browser .
The browser which got connected successfully I couldn't see such types 
of logs  .


What causes this header version in the wss connection ?


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


On Wed, Mar 20, 2024 at 6:51 PM Bogdan-Andrei Iancu 
 wrote:


Hi,

OpenSIPS does not like that 1.1 version in the VIA hdr - it
expects 2.0 (the SIP version).

Regards,

    Bogdan-Andrei Iancu

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

On 12.03.2024 10:57, Sasmita Panda wrote:

Hi All ,

1"  :callID 080a9ef8-3911-6c41-689c-fbb5d02e3aae
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via: bad char <1> on state 100
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via:  <1.1 p1wsa01.rivagecapital.com:80
<http://p1wsa01.rivagecapital.com:80>
(Cisco-WSA/12.5.4-011)#015#012#015#012>
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:parse_via: via parse failed
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:core:get_hdr_field: bad via
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
INFO:core:parse_headers: bad header field
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]:
ERROR:proto_wss:ws_parse_req_handshake: cannot parse
headers#012Upgrade: websocket#015#012Connection:
Upgrade#015#012Host: ir-webrtc2.3ccloud.com#015#012Pragma
<http://ir-webrtc2.3ccloud.com#015%23012Pragma>:
no-cache#015#012Cache-Control: no-cache#015#012User-Agent:
Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36
(KHTML, like Gecko) Chrome/122.0.0.0 <http://122.0.0.0>
Safari/537.36 Edg/122.0.0.0#015#012Origin
<http://122.0.0.0#015%23012Origin>:
https://ir-static.3ccloud.com#015#012Sec-WebSocket-Version
<https://ir-static.3ccloud.com#015%23012Sec-WebSocket-Version>:
13#015#012Accept-Encoding: gzip, br#015#012Accept-Language:
en-GB,en;q=0.9,en-US;q=0.8#015#012Sec-WebSocket-Key:
I/8XqtXyBCAGorZYjnTYQA==#015#012Sec-WebSocket-Extensions:
permessage-deflate;
client_max_window_bits#015#012Sec-WebSocket-Protocol:
sip#015#012X-IMForwards: 20#015#012Via: 1.1
p1wsa01.rivagecapital.com:80
<http://p1wsa01.rivagecapital.com:80>
(Cisco-WSA/12.5.4-011)#015#012#015#012


 I am getting this error on my opensips server where I am running
webrtc . What is this error for ?


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

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Re: [OpenSIPS-Users] call center

2024-03-20 Thread Bogdan-Andrei Iancu
Could you be more specific on your question, like what OpenSIPS should 
do for the call center application??


Regards,

Bogdan-Andrei Iancu

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

On 19.03.2024 07:03, Prathibha B wrote:

How to integrate Opensips with call center application?

--
Regards,
B.Prathibha

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Re: [OpenSIPS-Users] external applications

2024-03-20 Thread Bogdan-Andrei Iancu

Use the dialog events:
https://opensips.org/html/docs/modules/3.4.x/dialog.html#event_E_DLG_STATE_CHANGED

And you subscribe from outside OpenSIPS for such events:
https://www.opensips.org/Documentation/Interface-Events-3-4

Regards,

Bogdan-Andrei Iancu

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

On 20.03.2024 12:16, Prathibha B wrote:
No. I want to pass START, CONNECT, END messages from OpenSIPS to 
external application.


On Wed, 20 Mar 2024 at 15:42, Marcin Groszek  wrote:

Well, to execute external command from opensips you may want to
use EXEC module.

this is a manual for v3.2:

https://opensips.org/html/docs/modules/3.2.x/exec.html


On 3/20/2024 5:00 AM, Prathibha B wrote:

How to integrate OpenSIPS with external applications?

-- 
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B.Prathibha

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-- 
Best Regards:

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

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Re: [OpenSIPS-Users] TLS - Private Key fails

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

I see the root problem is OpenSIPS failing to load the private key file 
- I see some log about password case. So, when you generated the key, 
have you set a password for it?


Regards

Bogdan-Andrei Iancu

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

On 15.03.2024 19:35, Rodrigo R Passos wrote:


Hi everyone,


I'm trying to start opensips with a certificate generated by 
opensips-cli, like a tutorial. I have the following results keys after 
completing the command.


INFO: user certificate created in /etc/opensips/tls/user/wss-cert.pem

INFO: user private key created in /etc/opensips/tls/user/wss-privkey.pem

INFO: user CA list (chain of trust) created in 
/etc/opensips/tls/user/wss-calist.pem



This is my configuration, in opensips.cfg:

modparam("tls_mgm", "server_domain", "wss")
modparam("tls_mgm", "match_ip_address","[wss]10.0.0.114:5443 
<http://10.0.0.114:5443>")

modparam("tls_mgm", "ca_dir", "[wss]/etc/opensips/tls/rootCA/certs")
modparam("tls_mgm", "ca_list", 
"[wss]/etc/opensips/tls/user/wss-calist.pem")
modparam("tls_mgm", 
"certificate","[wss]/etc/opensips/tls/user/wss-cert.pem")
modparam("tls_mgm", 
"private_key","[wss]/etc/opensips/tls/user/wss-privkey.pem")

modparam("tls_mgm", "verify_cert", "[wss]0")
modparam("tls_mgm", "require_cert", "[wss]0")
modparam("tls_mgm", "ciphers_list", "[wss]NULL")
modparam("tls_mgm", "tls_method", "[wss]TLSv1")


So, when I start opensips, the log shows information saying the 
private key doesn't match with the public key. Follow the log:



Mar 15 14:27:14 esbc01 opensips[80845]: Mar 15 14:27:14 [80845] 
ERROR:tls_openssl:load_private_key: unable to load private key file 
'/etc/opensips/tls/user/wss-privkey.pem'.


Mar 15 14:27:14 esbc01 opensips[80845]: Retry (2 left) (check password 
case)


Mar 15 14:27:14 esbc01 opensips[80845]: Mar 15 14:27:14 [80845] 
ERROR:tls_openssl:load_private_key: key 
'/etc/opensips/tls/user/wss-privkey.pem' does not match the public key 
of the certificate


Mar 15 14:27:14 esbc01 opensips[80845]: Mar 15 14:27:14 [80845] 
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'wss'


Mar 15 14:27:14 esbc01 opensips[80845]: Mar 15 14:27:14 [80845] 
ERROR:core:init_mod: failed to initialize module tls_mgm


Mar 15 14:27:14 esbc01 opensips[80845]: Mar 15 14:27:14 [80845] 
ERROR:core:main: error while initializing modules


All procedures to create my own certificate follow the tutorial.


Anyone know what is happening or have any problems like me?


BR


RP



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Re: [OpenSIPS-Users] Freeswitch integration

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

Who's generating the 407 reply? OpenSIPS or FS ?

Regards,

Bogdan-Andrei Iancu

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

On 15.03.2024 13:03, Dragan, Calin via Users wrote:


Hi,

I’m trying to use Freeswitch as a media server with Opensips as a load 
balancer, I need something very basic for a POC. (I configured the 
load balancer and dispatcher based on old information found here: 
https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Enterprise-Deployment/Enterprise-deployment-OpenSIPS_13173169/#freeswitch-configuration, 
here: 
https://developer.signalwire.com/freeswitch/FreeSWITCH-Explained/Auxiliary-Knowledge-and-Utilities/OpenSIPS-configuration-for-2-or-more-FreeSWITCH-installs_13173462/ 
and here: 
https://www.opensips.org/Documentation/Tutorials-OpenSIPSFreeSwitchIntegration)


I’m adding the X-Auth-IP header to the requests like this (this is 
based on the config that ships with the 3.4.x release in the docker 
image, so I guessed that this is where I should add the header):


route[relay] {

   # for INVITEs enable some additional helper routes

   if (is_method("INVITE")) {

 remove_hf("X-Auth-IP");

append_hf("X-Auth-IP: $si\r\n");

append_to_reply("X-Auth-IP: $si\r\n");

xlog("L_INFO", "$ci|log|X-Auth-IP: $si\n");

xlog("L_INFO", "SIP Message: $mb");

t_on_failure("missed_call");

t_on_branch("per_branch_ops");

t_on_reply("handle_nat");

   }

   if (!t_relay()) {

send_reply(500,"Internal Error");

   }

   exit;

}

On the freeswitch side I added the IP set by opensips in the list of 
allowed addresses.


Now, with or without authentication enabled on the Freeswitch side 
(i.e. regardless of what I set in the “auth-calls” param, or if I 
enable the accept-blind-auth), or even whether the Freeswitch is 
running or not, when I try to place a call from/to some registerd 
addresses, I get the “407 Proxy authentication required” response.


Could you please give me some hints on what I might be doing wrong, or 
direct me to a contemporary tutorial?



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Re: [OpenSIPS-Users] Management Interface

2024-03-20 Thread Bogdan-Andrei Iancu

OpenSIPS Control Panel
https://controlpanel.opensips.org

Regards

Bogdan-Andrei Iancu

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

On 19.03.2024 09:30, Prathibha B wrote:

What is the alternative to Asterisk Management Interface in OpenSIPS?

--
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B.Prathibha

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Re: [OpenSIPS-Users] push notifications - and now packages failing

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

Just use rtpproxy or rtpengine instead, if not able to install the 
mediapoxy package.


Regards,

Bogdan-Andrei Iancu

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

On 15.03.2024 12:32, r...@rvgeerligs.nl wrote:

Hi all,

Maybe I thought wrong and I tried another system with AMD-INTEL 
architecture.


Desperate I started to install packages.


The following packages have unmet dependencies:

mediaproxy-common : Depends: python3 (< 3.9) but 3.10.6-1~22.04 is to 
be installed


E: Unable to correct problems, you have held broken packages.



Any advice is appreciated

Regards,

Ronald


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Re: [OpenSIPS-Users] bad uri, state 0 parsed: <$avp> (4) / <$avp(r)> (7)

2024-03-20 Thread Bogdan-Andrei Iancu

Hi

drop the quotes in :
    $avp(vals) = "$avp(r)";

as you want to put in $avp(vals) the VALUE of the $avp(r) and not the 
STRING "$avp(r)", I guess


Regards,

Bogdan-Andrei Iancu

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

On 14.03.2024 19:56, Social Boh wrote:

Hello,

any hint with this error:

ERROR:core:parse_uri: bad uri, state 0 parsed: <$avp> (4) / <$avp(r)> (7)

The lines are:

$avp(params) = "ruri";
$avp(vals) = "$avp(r)";:

on onreply route

$avp(r) contain the request uri created on the failure route.

Regards




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Re: [OpenSIPS-Users] how to debug many dialogs stuck in state 5?

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

What OpenSIPS version are you using? and what module do you use on top 
of the dialog module?


Regards,

Bogdan-Andrei Iancu

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

On 13.03.2024 13:55, Babak Yakhchali wrote:

Hi
When calling and immediately cancelling, the call is ended but 
dlg_list shows the dialog stuck in state 5. Increasing log level to 4 
shows these messages:
Mar 13 15:15:37 : DBG:tm:timer_routine: timer 
routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=26

Mar 13 15:15:37 : DBG:tm:delete_handler: removing 0x7f1c861d0630
Mar 13 15:15:37 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: 
can't delete -- still reffed (-1)

Mar 13 15:15:37 : DBG:tm:set_timer: relative timeout is 2
Mar 13 15:15:37 : DBG:tm:insert_timer_unsafe: [3]: 0x7f1c861d06e0 (28)
Mar 13 15:15:37 : DBG:tm:delete_handler: done
Mar 13 15:15:39 : DBG:tm:timer_routine: timer 
routine:3,tl=0x7f1c861d06e0 next=(nil), timeout=28

Mar 13 15:15:39 : DBG:tm:delete_handler: removing 0x7f1c861d0630
Mar 13 15:15:39 : DBG:tm:delete_cell: delete_cell 0x7f1c861d0630: 
can't delete -- still reffed (-1)

Mar 13 15:15:39 : DBG:tm:set_timer: relative timeout is 2
Mar 13 15:15:39 : DBG:tm:insert_timer_unsafe: [3]: 0x7f1c861d06e0 (30)
Mar 13 15:15:39 : DBG:tm:delete_handler: done

How can I debug the issue? What are the possible causes of this?
thanks

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Re: [OpenSIPS-Users] utimer_ticker warning in opensips 3.2

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

How the two reports fit together here ? there are completely separate 
experiences on different runs?? or if you start opensips first you get 
the warnings and later it crashes ??
For the crash part, I see a core file was generated - could you extract 
the backtrace and post here ? (see 
https://opensips.org/Documentation/TroubleShooting-Crash)


Regards

Bogdan-Andrei Iancu

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

On 11.03.2024 15:04, Sasmita Panda wrote:

Any update on this ?


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


On Mon, Mar 11, 2024 at 12:03 PM Sasmita Panda  wrote:

With the same server configuration and opensips version I am
getting below error as well .

CRITICAL:core:fm_free: freeing already freed shm pointer
(0x7fc110e0b408), first free: (null): (null)(0) - aborting!

 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs: child
process 215185 exited by a signal 6
 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs: core was
generated
 /usr/local/sbin/opensips[215171]: INFO:core:handle_sigs:
terminating due to SIGCHLD
 /usr/local/sbin/opensips[215172]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: fm_status (0x7fc2907ff010):
 /usr/local/sbin/opensips[215174]: INFO:core:sig_usr: signal 15
received
 /usr/local/sbin/opensips[215172]: heap size= 33554432
 /usr/local/sbin/opensips[215174]: Memory status (pkg):
 /usr/local/sbin/opensips[215172]: used= 3710048,
used+overhead=3801344, free=29844384
 /usr/local/sbin/opensips[215174]: fm_status (0x7fc2907ff010):
 /usr/local/sbin/opensips[215172]: max used (+overhead)= 3801344
 /usr/local/sbin/opensips[215174]: heap size= 33554432
 /usr/local/sbin/opensips[215172]: dumping free list:
 /usr/local/sbin/opensips[215174]: used= 229752,
used+overhead=318936, free=33324680
 /usr/local/sbin/opensips[215172]: hash =   1 fragments no.:    
1, unused:     0#012#011#011 bucket size: 8 -         8 (first    
    8)
 /usr/local/sbin/opensips[215174]: max used (+overhead)= 385792
 /usr/local/sbin/opensips[215174]: dumping free list:
 /usr/local/sbin/opensips[215172]: hash =  18 fragments no.:    
1, unused:     0#012#011#011 bucket size: 144 -       144 (first  
    144)
 /usr/local/sbin/opensips[215174]: hash =   7 fragments no.:  
139, unused:     0#012#011#011 bucket size:  56 -        56 (first
       56)
 /usr/local/sbin/opensips[215172]: hash = 2059 fragments no.:    
1, unused:     0#012#011#011 bucket size:  16777216 -  33554432
(first  29752936)
 /usr/local/sbin/opensips[215174]: hash =  13 fragments no.:  
 37, unused:     0#012#011#011 bucket size: 104 -       104 (first
      104)
 /usr/local/sbin/opensips[215172]: TOTAL:      3 free fragments =
29753088 free bytes
 /usr/local/sbin/opensips[215172]: TOTAL: 48 overhead
 /usr/local/sbin/opensips[215174]: hash =  16 fragments no.:  
 61, unused:     0#012#011#011 bucket size: 128 -       128 (first
      128)
 /usr/local/sbin/opensips[215172]: -
 /usr/local/sbin/opensips[215174]: hash =  31 fragments no.:  
152, unused:     0#012#011#011 bucket size: 248 -       248 (first
      248)
 /usr/local/sbin/opensips[215174]: hash =  68 fragments no.:  
 16, unused:     0#012#011#011 bucket size: 544 -       544 (first
      544)
 /usr/local/sbin/opensips[215174]: hash = 105 fragments no.:    
1, unused:     0#012#011#011 bucket size: 840 -       840 (first  
    840)
 /usr/local/sbin/opensips[215174]: hash = 2059 fragments no.:    
1, unused:     0#012#011#011 bucket size:  16777216 -  33554432
(first  33168816)
 /usr/local/sbin/opensips[215174]: TOTAL:    407 free fragments =
33235496 free bytes
 /usr/local/sbin/opensips[215174]: TOTAL: 48 overhead
 /usr/local/sbin/opensips[215174]: -
 /usr/local/sbin/opensips[215176]: Memory status (pkg):
 /usr/local/sbin/opensips[215176]: fm_status (0x7fc2907ff010):
 /usr/local/sbin/opensips[215176]: heap size= 33554432
 /usr/local/sbin/opensips[215176]: used= 3705480,
used+overhead=3796584, free=29848952
 /usr/local/sbin/opensips[215176]: max used (+overhead)= 3798064
 /usr/local/sbin/opensips[215176]: dumping free list:
 /usr/local/sbin/opensips[215176]: hash =   2 fragments no.:    
2, unused:     0#012#011#011 bucket size:  16 -        16 (first  
     16)
 /usr/local/sbin/opensips[215176]: hash =   3 fragments no.:    
1, unused:     0#012#011#011 bucket size:  24 -        24 (first  
     24)
 /usr/local/sbin/opensips[215176]: hash =  16 fragments no.:    
1, unused:     0#012#011#011 bucket size: 128 -       128 (first  
    128)
 /usr/local/

Re: [OpenSIPS-Users] SDP

2024-03-20 Thread Bogdan-Andrei Iancu
Nothing is impossible as time as you search in the right place (aka the 
Manual). See my reply on your thread.


Regards,

Bogdan-Andrei Iancu

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

On 14.03.2024 09:50, Stefan Carlsson wrote:


Hi !

You can use the module sipmsgops to view and change some of the ADP 
fields such as codecs and streams … but unfortunately you don’t have


complete control of the SDP compared to what it seems could be done on 
the Kamailio SIP router


I need to change the a=sendrecv field and it seems to be impossible ☹

Regards …

_

Stefan Carlsson

*From:* Users  *On Behalf Of 
*Prathibha B

*Sent:* Thursday, 14 March, 2024 05:16
*To:* OpenSIPS users mailling list 
*Subject:* [OpenSIPS-Users] SDP

How to see the SDP in opensips?

--

Regards,

B.Prathibha


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Re: [OpenSIPS-Users] SDP

2024-03-20 Thread Bogdan-Andrei Iancu

The online Manual should be your Bible here ;)

https://opensips.org/Documentation/Script-CoreVar-3-4#rb

Regards,

Bogdan-Andrei Iancu

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

On 14.03.2024 06:16, Prathibha B wrote:

How to see the SDP in opensips?

--
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B.Prathibha

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Re: [OpenSIPS-Users] How to change the a=sendrecv in SDP

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

See
https://opensips.org/html/docs/modules/3.4.x/textops.html#func_replace_body_all 


or
https://opensips.org/html/docs/modules/3.4.x/textops.html#func_subst_body

Regards

Bogdan-Andrei Iancu

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

On 13.03.2024 17:19, Stefan Carlsson wrote:


Hi (again …)

I’ve found out how to strip out the o= to separate values …  but now I 
want to change the a=sendrecv to a=sendonly


The cause is  that our pbx’s don’t conform to either RFC2543 hold 
type: null connection IP detected,  nor RFC3264 hold type: inactive or 
sendonly attributes detected


but use the RFC5359 re-invite indicating call hold via session version 
incremented by  1.


So now I’ve trying to make a workaround with the help of OpenSips ..

Can you please help me with that …

// Regards …

Stefan Carlsson


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Re: [OpenSIPS-Users] Regex help.

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

Use the s.subst [1] transformation to replace any continues set of 
spaces with a semi-colon


[1] https://www.opensips.org/Documentation/Script-Tran-3-4#re.subst

Regards,

Bogdan-Andrei Iancu

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

On 12.03.2024 19:18, Stefan Carlsson wrote:


Hi !

What is the easiest form to transform this:

o=SYS   1710262255    1710262255    IN    IP4 123.123.123.123

Into

SYS;1710262255;1710262255;IN;IP4;123.123.123.123

for later retrieval via an index function.

[0] = SYS

[1] = 1710262255

[2] = 1710262255

[3] = IN

etc ….

I need to get my hands on the session parameters …

Thanks in advance …

//S


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Re: [OpenSIPS-Users] OPENSIP 3.4 send_reply and tight matching failed

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

The script snippet has nothing to do with the matching - it is about a 
200-ok ACK to be matched by the dialog module and the TO-tag does not 
match at all.


The link to the trace expired, if you could repost.

Regards,

Bogdan-Andrei Iancu

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

On 12.03.2024 14:57, Alain Bieuzent wrote:


Hi all,

I am in the process of migrating my proxies from 3.3.9 to 3.4.4.

In 3.4 I find myself with a fault when I try to rewrite a final cause.

if (t_check_status("500"))

{

send_reply(503,"Service Unavailable");

exit;

  }

with 3.3 version, this code works perfectly, since 3.4 I have this error:

WARNING:dialog:dlg_onroute: tight matching failed for ACK with 
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52, 
ftag='as5328cea7'/10, ttag='b5a7-2a346bb9d510da893a ee7cda850584cb'/37 
and direction=1


WARNING:dialog:dlg_onroute: dialog identification elements are 
callid='67457c401e9ea52e2750100979253a7a@10.101.180.177:5060'/52, 
caller tag='as5328cea7'/10, callee tag='3919223738-770090972'/ 20


Has anyone encountered this error before?

Complete traces here : https://www.transfernow.net/dl/20240312qvAFfIAL.

Regards


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Re: [OpenSIPS-Users] Need some help on opensips error for wss handshek .

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

OpenSIPS does not like that 1.1 version in the VIA hdr - it expects 2.0 
(the SIP version).


Regards,

Bogdan-Andrei Iancu

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

On 12.03.2024 10:57, Sasmita Panda wrote:

Hi All ,

1"  :callID 080a9ef8-3911-6c41-689c-fbb5d02e3aae
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
ERROR:core:parse_via: bad char <1> on state 100
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
ERROR:core:parse_via:  <1.1 p1wsa01.rivagecapital.com:80 
<http://p1wsa01.rivagecapital.com:80> 
(Cisco-WSA/12.5.4-011)#015#012#015#012>
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
ERROR:core:parse_via: via parse failed
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
ERROR:core:get_hdr_field: bad via
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
INFO:core:parse_headers: bad header field
Mar 12 08:23:11 localhost /usr/sbin/opensips[12987]: 
ERROR:proto_wss:ws_parse_req_handshake: cannot parse 
headers#012Upgrade: websocket#015#012Connection: Upgrade#015#012Host: 
ir-webrtc2.3ccloud.com#015#012Pragma 
<http://ir-webrtc2.3ccloud.com#015#012Pragma>: 
no-cache#015#012Cache-Control: no-cache#015#012User-Agent: Mozilla/5.0 
(Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) 
Chrome/122.0.0.0 <http://122.0.0.0> Safari/537.36 
Edg/122.0.0.0#015#012Origin <http://122.0.0.0#015#012Origin>: 
https://ir-static.3ccloud.com#015#012Sec-WebSocket-Version 
<https://ir-static.3ccloud.com#015#012Sec-WebSocket-Version>: 
13#015#012Accept-Encoding: gzip, br#015#012Accept-Language: 
en-GB,en;q=0.9,en-US;q=0.8#015#012Sec-WebSocket-Key: 
I/8XqtXyBCAGorZYjnTYQA==#015#012Sec-WebSocket-Extensions: 
permessage-deflate; 
client_max_window_bits#015#012Sec-WebSocket-Protocol: 
sip#015#012X-IMForwards: 20#015#012Via: 1.1 
p1wsa01.rivagecapital.com:80 <http://p1wsa01.rivagecapital.com:80> 
(Cisco-WSA/12.5.4-011)#015#012#015#012



 I am getting this error on my opensips server where I am running 
webrtc . What is this error for ?



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

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Re: [OpenSIPS-Users] pn still problems

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

In the log snippet, I do not see any of the xlogs you have after the 
lookup() 


Regards

Bogdan-Andrei Iancu

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

On 09.03.2024 15:46, r...@rvgeerligs.nl wrote:

Hi All,

I still have problems with opensips3.4 push notifications:

Somehow INVITEs keep looping and get rejected.

Code:
...
route{

...

if (has_totag()) {

...

if (!is_method("ACK"))
    async (pn_process_purr("location"), resume_route);

...
}

...

# do lookup with method filtering

#--
lookup("location", "method-filtering");
$var(rc) = $retcode;
switch ($var(rc)) {
case 1:
    # we found at least 1 non-PN contact!
xlog("L_INFO", "DBG: 1 contacts found non-PN ($var(rc))\n");
    $var(do_relay) = true;
    break;
case 2:
    # success, but all contacts are PN-enabled, so we're
    # sending PNs / awaiting re-registrations from them
xlog("L_INFO", "DBG: contacts found send PN? ($var(rc))\n");
    $var(do_relay) = false;
    #$avp(skip_relay) = true;
    break;
default:
    xlog("L_INFO", "DBG: no contacts found ($var(rc))\n");
    t_reply(404, "Not Found");
    exit;
}

if ($var(do_relay) && !t_relay())
    send_reply(500, "Internal Server Error");

...

if($var(do_relay) == true)
route(relay);
}

...

from log (looping log, it repeats):

DBG:core:parse_headers: this is the first via
2024-03-09T09:41:13.485706+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:_parse_to: end of header 
reached, state=10
2024-03-09T09:41:13.485748+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:_parse_to: display={}, 
ruri={sip:t...@dummy.com}
2024-03-09T09:41:13.485807+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_hdr_field:  [20]; 
uri=[sip:t...@dummy.com]
2024-03-09T09:41:13.485854+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_hdr_field: to body 
[#015#012]
2024-03-09T09:41:13.485915+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_hdr_field: cseq 
: <1>
2024-03-09T09:41:13.485962+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_hdr_field: found end 
of header
2024-03-09T09:41:13.486040+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:parse_headers: 
flags=
2024-03-09T09:41:13.486088+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_dummy_sip_msg: setting 
as static to 0xb1a612c0
2024-03-09T09:41:13.486145+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:evi_raise_event_msg: found 
subscriber E_UL_CONTACT_REFRESH
2024-03-09T09:41:13.486207+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: 
DBG:core:dup_ref_script_route_in_shm: dupping 0xaa264f60 
[E_UL_CONTACT_REFRESH], idx 1, ver/cnt 1, to new 0xaa26d9e0 
[E_UL_CONTACT_REFRESH], idx 1, ver/cnt 1
2024-03-09T09:41:13.486283+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:release_dummy_sip_msg: 
cleaning the static sip msg 0xb1a612c0
2024-03-09T09:41:13.486352+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:destroy_avp_list: 
destroying list (nil)
2024-03-09T09:41:13.486409+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:ipc_handle_job: received 
job type 0[RPC] from process 6
2024-03-09T09:41:13.486467+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:get_dummy_sip_msg: reusing 
the static sip msg 0xb1a612c0
2024-03-09T09:41:13.486533+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:parse_params: Parsing 
params 
for:[sip:rvg@82.151.164.77:57775;pn-prid=F1B7F53A99A35254A55CCAE2EDF80C1FAE12F353592BB6ED203ADA0175E34AD9:voip;pn-provider=apns;pn-param=ABCD1234.cllkit.voip;pn-silent=1;pn-timeout=0;transport=udp]
2024-03-09T09:41:13.486571+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: ERROR: -- ** --pn_send was called 
sip:u...@dummy.com
2024-03-09T09:41:13.486651+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: ERROR:core:route_param_get: route 
$param variable accepts only integer indexes
2024-03-09T09:41:13.486746+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: message repeated 2 times: [ 
ERROR:core:route_param_get: route $param variable accepts only integer 
indexes]
2024-03-09T09:41:13.486787+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: ERROR:pn params: 
,#012   ,#012
2024-03-09T09:41:13.486822+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:release_dummy_sip_msg: 
cleaning the static sip msg 0xb1a612c0
2024-03-09T09:41:13.486913+00:00 xsp2 
/usr/local/sbin/opensips[2061136]: DBG:core:destroy_avp_list: 
destroying list (nil)
2024-03-09T09:41:13.697168+00:00 xsp2 
/usr/local/sbin/opensips[2061135]: DBG:core:parse_msg: SIP Request:


Any advice appreciated, regards,

Ronald Geerligs






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Re: [OpenSIPS-Users] Problems with new installation of 3.4

2024-03-20 Thread Bogdan-Andrei Iancu

Hi,

Some opensips modules are provided via separate debs (due the 
dependencies), so you just have to install more packages, no need to 
compile anything:

https://apt.opensips.org/pool/jammy/3.4-releases/o/opensips/

What you need here is opensips-auth-modules, I guess

Regards,

Bogdan-Andrei Iancu

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

On 15.03.2024 20:27, Stefan Carlsson wrote:


Hi !

After followed the installation of 3.4 LTS to a fresh ubuntu ver 22 a 
found some serious problems …


installing is done via apt.

1) Not all the modules were included in the rep. as it should ref. to 
spec. vers.  manual


2) After downloading the source , trying to compile the modules 
missing…  dependencies needed and info is missing in the main doc. or 
moduledoc.


For example:

make: *** No rule to make target 
'../../lib/digest_auth/libdigest_auth.a', needed by 'uac_auth.so'.  Stop.


trying to make all from opensips src. root folder …

Compiling main.c

In file included from main.c:152:

ssl_tweaks.h:2:10: fatal error: openssl/opensslv.h: No such file or 
directory


Any ideas …

Mvh // Regards …

_

Stefan Carlsson


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[OpenSIPS-Users] [Blog] The SQL support, reloaded (in OpenSIPS 3.5)

2024-03-13 Thread Bogdan-Andrei Iancu


Born from the AVPops ashes, the SQLops module is here to provide JSON 
structured SQL querying capabilities, from script level, with prepared 
statements support and many other facilities and boosts . Read all 
about this


    https://blog.opensips.org/2024/03/13/the-sql-support-reloaded/

Enjoy,

--
Bogdan-Andrei Iancu

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


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Re: [OpenSIPS-Users] possible big in unixodbc queries

2024-03-05 Thread Bogdan-Andrei Iancu
The data type of the each column is provided by the unixodbc lib via the 
SQLDescribeCol() function.


I would rather suspect an issue between the unixodbc layer and the 
underlying DB.


Regards,

Bogdan-Andrei Iancu

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

On 02.03.2024 02:57, Saint Michael wrote:

whatever query I call, db_unixodbc_get_columns returns only db_int for
every field.
DBG:avpops:ops_dbquery_avps: query [CALL get_user_details(1)]
  DBG:core:db_new_result: allocate 48 bytes for result set at 0x71c49935ba20
  DBG:db_unixodbc:db_unixodbc_get_columns: 3 columns returned from the query
  DBG:core:db_allocate_columns: allocate 84 bytes for result columns at
0x71c49935ba88
  DBG:db_unixodbc:db_unixodbc_get_columns:
RES_NAMES(0x71c49935baa0)[0]=[username]
  DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
  DBG:db_unixodbc:db_unixodbc_get_columns: RES_NAMES(0x71c49935bab0)[1]=[email]
  DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
  DBG:db_unixodbc:db_unixodbc_get_columns: RES_NAMES(0x71c49935bac0)[2]=[phone]
  DBG:db_unixodbc:db_unixodbc_get_columns: use DB_INT result type
  DBG:core:db_allocate_rows: allocate 112 bytes for result rows and
values at 0x71c49935bb18
  DBG:db_unixodbc:db_unixodbc_str2val: converting INT [johndoe]
  ERROR:core:db_str2int: Unexpected characters: [johndoe]
  ERROR:db_unixodbc:db_unixodbc_str2val: converting integer value from
string failed
  ERROR:db_unixodbc:db_unixodbc_convert_row: failed to convert value

opensips -V
version: opensips 3.4.4 (x86_64/linux)
flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC,
Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll, sigio_rt, select.
git revision: e50c6736e
main.c compiled on 00:29:53 Mar  2 2024 with gcc 9

dpkg -l | grep unixodbc
ri  unixodbc 2.3.6-0.1build1
amd64Basic ODBC tools
ri  unixodbc-dev:amd64   2.3.6-0.1build1
amd64ODBC libraries for UNIX (development files)

odbc.ini

[opensips]
Description = MySQL ODBC Driver Testing
Driver = maria
Socket = /var/run/mysqld/mysqld.sock
#Server = localhost
User = x
Password = 
Database = asterisk
Option = 3

odbcinst.ini

[maria]
Description=ODBC for MySQL
Driver=/usr/lib64/mariadb/libmaodbc.so
FileUsage=1
Threading=0

a Unixodbc trace shows that everything is normal,so the issue must be
with opensips
 Statement = 0x6003034831b0
 Column Number = 3
 Target Type = 1 SQL_CHAR
 Buffer Length = 1024
 Target Value = 0x77a24e1ff318
 StrLen Or Ind = 0x7ffc233ad5f0
[ODBC][103886][1709340852.301423][SQLGetData.c][545]
 Exit:[SQL_SUCCESS]
 Buffer = [123-456-7890]
 Strlen Or Ind = 0x7ffc233ad5f0 -> 1

as you can see, the target Type is SQL_CHAR

any ideas?

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[OpenSIPS-Users] [Blog] Launch Darkly integration with OpenSIPS 3.5

2024-03-04 Thread Bogdan-Andrei Iancu

Hi all,

Starting version 3.5 (current devel), OpenSIPS offers integration with 
Launch Darkly service for in-cloud features flag support


https://blog.opensips.org/2024/03/04/launchdarkly-integration-with-opensips-3-5/

Enjoy,

--
Bogdan-Andrei Iancu

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


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Re: [OpenSIPS-Users] Via header IP

2024-02-28 Thread Bogdan-Andrei Iancu

Hi Jason,

Not sure about that, as time as I'm not sure about the misbehavior here. 
Usually the "advertising" per message is tricky to use because there is 
no persistence (across sip packages) of the advertising you did on the 
request.


The best / safer way to go is with advertising per socket/listener, if 
possible.


Regards,

Bogdan-Andrei Iancu

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

On 26.02.2024 22:36, nz deals wrote:

Hi Bogdan,

Do you believe that removing the 'advertised_address' from the global 
configuration will solve the issue?




Regards,
Jason

On Sat, 24 Feb 2024 at 10:09, nz deals  wrote:

Here it is
version: opensips 3.4.2 (x86_64/linux)
git revision: 8a6e78dde


Regards,
Jason

On Sat, 24 Feb 2024 at 01:53, Bogdan-Andrei Iancu
 wrote:

What version / revision (opensips -v) do you have there ?

Regards,

    Bogdan-Andrei Iancu

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

On 23.02.2024 14:02, nz deals wrote:

Hi Bogdan,

The opensips is only listening on 192.xx. internal ip. It is
Google VM so the public IP 202.x is assigned.

Yes, that is correct. I advertise from a script
like set_advertised_address("192.x")
but I have the following in the config.

advertised_address=202.x
alias=202.x

Yes, I use t_relay for INVITE and CANCEL.

Just more detail for CANCEL. I use the following:-
if (is_method("CANCEL")) {
                if (t_check_trans())
                t_relay();
                exit;
}

Thanks

Regards,
Jason

On Sat, 24 Feb 2024 at 00:53, Bogdan-Andrei Iancu
 wrote:

Hi Jason

So you advertise fro m script the 192.x address and also
have the global
advertise with 202.x ? is you opensips listening on a
totally different IP?

And for the INVITE and CANCEL, you use t_relay(), right ?

    Regards,

    Bogdan-Andrei Iancu

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

On 22.02.2024 23:08, nz deals wrote:
> Hello everyone,
>
> I have a weird issue, when i send the invite i force
the following.
> set_advertised_address("192.168.XX.XX");
> so it is using my internal IP in the VIA.
>
> When i send CANCEL, it is using my Public IP which is
also defined in
> the config as a global. advertised_address=202.xx.xx.xx
> alias= 202.xx.xx.xx
>
> Anyway to fix my CANCEL's VIA so it starts using my
internal ip i.e
> 192.168.xx.xx ?
> I tried it manually via setting set_advertised but it
does not get
> changed on CANCEL.
>
> Thanks.
>
> Regards,
> Jason
>
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

2024-02-26 Thread Bogdan-Andrei Iancu
yes, you can use the b2b_logic (together with b2b_entities) for that, 
but it may be a too heavy tool for the purpose. Maybe you can try to 
generate the PRACK from OpenSIPS level by using the t_uac_dlg MI 
function [1] via the mi_script module [2] - basically to trigger that MI 
cmd from the onreply_route, when receiving the 180 reply.


[1] https://opensips.org/html/docs/modules/3.4.x/tm.html#mi_t_uac_dlg
[2] https://opensips.org/html/docs/modules/3.4.x/mi_script.html#afunc_mi

Regards,

Bogdan-Andrei Iancu

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

On 25.02.2024 14:54, Social Boh wrote:


I know the best way is the original caller handle and reply the 183, 
but I HAVE to do this at OpenSIPs level.


Can I use B2B_Entities and B2B_logic to do this? Is there a scenario 
to use tu parse and reply the 183?


Thank you

---
I'm SoCIaL, MayBe
El 16/02/2024 a las 11:31 a. m., Bogdan-Andrei Iancu escribió:
I agree here, the 183 must be relayed back to the original caller 
(which generate the received INVITE) and let it do the PRACK - this 
confirmation must be end-2-end in the dialog.


Regards,
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 12.02.2024 11:36, Ihor Olkhovskyi wrote:
You should relay 183 to original source (ione that is sending 
INVITE) and got PRACK from there.

That would be the most correct way of handling this

Le lun. 12 févr. 2024 à 02:29, Social Boh  a écrit :

Maybe someone can help me.

This is the scenario.

OpenSIPs receive a INVITE and send it to a server that reply
with a 302
message (always)

Then OpenSIPs, in the failure route, take the user part present
in the
302 contact header, change the destination IP and send with t_relay

The destination reply with a 183 with Require: 100rel header so
OpenSIPs
have to reply with a PRACK. This is my problem.

I don't know which is the best way to handle this (the PRACK)

Thank you

Regards

---
I'm SoCIaL, MayBe

El 9/02/2024 a las 6:46 a. m., Social Boh escribió:
> Hello list,
>
> can OpenSIPs generate a PRACK message to reply a 180/183 message?
>
> Thank you
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
>
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--
Best regards,
Ihor (Igor)

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Re: [OpenSIPS-Users] infinite loop in rtpengine rtpe_function_call()

2024-02-26 Thread Bogdan-Andrei Iancu

Thanks Alexander, let's continue over there.

Regards,

Bogdan-Andrei Iancu

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

On 24.02.2024 17:17, Alexander Kogan wrote:


Hi,

it's submitted as https://github.com/OpenSIPS/opensips/issues/3314

Best regards,
Alexander Kogan,
Director of R
5g Future
http://5gfuture.com
On 23.02.2024 15:56, Bogdan-Andrei Iancu wrote:

Hi Alexander,

Thank for your report - it will be better to push this via the github 
tracker, the right place for coding issues:

https://github.com/OpenSIPS/opensips/issues
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.com
   https://www.siphub.com
On 22.02.2024 21:43, Alexander Kogan wrote:


Hi,

The OpenSIPS process gets stack in infinite loop when the rtpengine 
module runs out of memory in send_rtpe_command()


Please review the following piece of code in rtpengine.c:

    RTPE_START_READ();
    do {
    if (snode && snode->s) {
    if ((node = get_rtpe_node(snode, set)) == NULL && op == 
OP_OFFER)

    node = select_rtpe_node(ng_flags.call_id, 1, set);
    snode = NULL;
    } else {
    node = select_rtpe_node(ng_flags.call_id, 1, set);
    }
    if (!node) {
    LM_ERR("no available proxies\n");
    RTPE_STOP_READ();
    goto error;
    }

    cp = send_rtpe_command(node, ng_flags.dict, );
    } *while (cp == NULL);*
    RTPE_STOP_READ();
    LM_DBG("proxy reply: %.*s\n", ret, cp);

When send_rtpe_command() returns NULL due to memory absence here:

    v = bencode_iovec(dict, , 1, 0);
    if (!v) {
    LM_ERR("error converting bencode to iovec\n");
    return NULL;
    }

the mentioned loop ( *do {} while(cp == NULL)* ) won't stop 
indefinitely causing the opensips process to become stuck.


Best regards,
Alexander Kogan,
Director of R
5g Future
http://5gfuture.com

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Re: [OpenSIPS-Users] unicode characters in dlg_list caller_contact field on re-invite

2024-02-26 Thread Bogdan-Andrei Iancu

Denis,

I do not see any unicode chars to be present in the received contact 
URIs. What you see in the dlg_list output looks like garbage to me - 
maybe the update was not properly done and the len of the contact is 
longer than the actual value, leading to to memory garbage to be printed 
(printing out of buffer).


Do you use from script any functions that may change the contact, like 
fix_nated_contact() or so ?


Regards,

Bogdan-Andrei Iancu

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

On 24.02.2024 05:35, Denis Lyubimov wrote:

Hi,

I will put here an example of what I get in caller_contact with re-INVITE.

This is incoming INVITE:

INVITE sip:7654321@192.168.224.50 
<mailto:sip%3A7654321@192.168.224.50>;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 
172.17.17.55:49677;rport;branch=z9hG4bKPj0c5e8cf59ad645119867f42d06c77a9f;alias

Max-Forwards: 70
From: <mailto:sip%3A1002@192.168.224.50>>;tag=b8b13e3376954c7dbaff8da68ee1d44a

To: mailto:sip%3A7654321@192.168.224.50>>
Contact: 
Call-ID: 5e97161dba344eeeab00d40bc883e4e3
CSeq: 17799 INVITE
Route: 


This is forwarded INVITE:

INVITE sip:7654321@192.168.224.50 
<mailto:sip%3A7654321@192.168.224.50>;transport=tcp SIP/2.0
Record-Route: 

68ee1d44a;nat=yes;did=61f.b845fdd7>
Record-Route: 

4a;nat=yes;did=61f.b845fdd7>
Via: SIP/2.0/TCP 
192.168.224.91:5090;branch=z9hG4bK86c6.385ee382.0;i=a51f1803
Via: SIP/2.0/TCP 
172.17.17.55:49677;received=4.4.4.1;rport=49677;branch=z9hG4bKPj0c5e8cf59

ad645119867f42d06c77a9f;alias
Max-Forwards: 69
From: <mailto:sip%3A1002@192.168.224.50>>;tag=b8b13e3376954c7dbaff8da68ee1d44a

To: mailto:sip%3A7654321@192.168.224.50>>
Contact: 
Call-ID: 5e97161dba344eeeab00d40bc883e4e3
CSeq: 17799 INVITE
...

dlg_list:

{
    "Dialogs": [
        {
            "ID": "61f.b845fdd7",
            "db_id": "16589275485323",
            "state": 4,
            "user_flags": "DISPATCHED_FLG",
            "timestart": 1708744350,
            "datestart": "2024-02-24 06:12:30",
            "timeout": 1708787549,
            "dateout": "2024-02-24 18:12:29",
            "callid": "5e97161dba344eeeab00d40bc883e4e3",
            "from_uri": "sip:1002@192.168.224.50 
<mailto:sip%3A1002@192.168.224.50>",
            "to_uri": "sip:7654321@192.168.224.50 
<mailto:sip%3A7654321@192.168.224.50>",

            "caller_tag": "b8b13e3376954c7dbaff8da68ee1d44a",
            "caller_contact": "sip:1002@4.4.4.1:49677;transport=TCP;ob",
            "callee_cseq": "0",
            ...

For now caller_contact has the value 
"sip:1002@4.4.4.1:49677;transport=TCP;ob".



Then i do re-INVITE from caller:

INVITE sip:192.168.224.50:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 
172.17.17.55:49677;rport;branch=z9hG4bKPjdfcc1bcad866428c8d43d1fbcfeaba54;alias

Max-Forwards: 70
From: <mailto:sip%3A1002@192.168.224.50>>;tag=b8b13e3376954c7dbaff8da68ee1d44a
To: <mailto:sip%3A7654321@192.168.224.50>>;tag=008162B2-5CFB-15D9-810C-32E0A8C0AA77-3

Contact: 
Call-ID: 5e97161dba344eeeab00d40bc883e4e3
CSeq: 17800 INVITE
...

forwarded re-INVITE:

INVITE sip:192.168.224.50:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 
192.168.224.91:5090;branch=z9hG4bKbb97.f5810342.0;i=a51f1803
Via: SIP/2.0/TCP 
172.17.17.55:49677;received=4.4.4.1;rport=49677;branch=z9hG4bKPjdfcc1bcad

866428c8d43d1fbcfeaba54;alias
Max-Forwards: 69
From: <mailto:sip%3A1002@192.168.224.50>>;tag=b8b13e3376954c7dbaff8da68ee1d44a
To: <mailto:sip%3A7654321@192.168.224.50>>;tag=008162B2-5CFB-15D9-810C-32E0A8C0AA77-3

Contact: 
Call-ID: 5e97161dba344eeeab00d40bc883e4e3
CSeq: 17800 INVITE
...

dlg_list:

{
    "Dialogs": [
        {
            "ID": "61f.b845fdd7",
            "db_id": "16589275485323",
            "state": 4,
            "user_flags": "DISPATCHED_FLG",
            "timestart": 1708744350,
            "datestart": "2024-02-24 06:12:30",
            "timeout": 1708787549,
            "dateout": "2024-02-24 18:12:29",
            "callid": "5e97161dba344eeeab00d40bc883e4e3",
            "from_uri": "sip:1002@192.168.224.50 
<mailto:sip%3A1002@192.168.224.50>",
            "to_uri": "sip:7654321@192.168.224.50 
<mailto:sip%3A76543

Re: [OpenSIPS-Users] Via header IP

2024-02-23 Thread Bogdan-Andrei Iancu

What version / revision (opensips -v) do you have there ?

Regards,

Bogdan-Andrei Iancu

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

On 23.02.2024 14:02, nz deals wrote:

Hi Bogdan,

The opensips is only listening on 192.xx. internal ip. It is Google VM 
so the public IP 202.x is assigned.


Yes, that is correct. I advertise from a script 
like set_advertised_address("192.x")

but I have the following in the config.

advertised_address=202.x
alias=202.x

Yes, I use t_relay for INVITE and CANCEL.

Just more detail for CANCEL. I use the following:-
if (is_method("CANCEL")) {
                if (t_check_trans())
                t_relay();
                exit;
}

Thanks

Regards,
Jason

On Sat, 24 Feb 2024 at 00:53, Bogdan-Andrei Iancu 
 wrote:


Hi Jason

So you advertise fro m script the 192.x address and also have the
global
advertise with 202.x ? is you opensips listening on a totally
different IP?

And for the INVITE and CANCEL, you use t_relay(), right ?

Regards,

    Bogdan-Andrei Iancu

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

On 22.02.2024 23:08, nz deals wrote:
> Hello everyone,
>
> I have a weird issue, when i send the invite i force the following.
> set_advertised_address("192.168.XX.XX");
> so it is using my internal IP in the VIA.
>
> When i send CANCEL, it is using my Public IP which is also
defined in
> the config as a global. advertised_address=202.xx.xx.xx
> alias= 202.xx.xx.xx
>
> Anyway to fix my CANCEL's VIA so it starts using my internal ip i.e
> 192.168.xx.xx ?
> I tried it manually via setting set_advertised but it does not get
> changed on CANCEL.
>
> Thanks.
>
> Regards,
> Jason
>
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Re: [OpenSIPS-Users] unicode characters in dlg_list caller_contact field on re-invite

2024-02-23 Thread Bogdan-Andrei Iancu

Hi Denis,

Could post the actual Contact hdr as received (network level like pcap 
or trace) by OpenSIPS from both caller and callee parties ?


Regards

Bogdan-Andrei Iancu

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

On 22.02.2024 07:34, Denis Lyubimov wrote:

Hi all,

My network scheme is
softphone <-nat-external network-nat-> opensips <-local network-> sip 
servers <-local network-> softphone

No b2b or topology_hiding modules are used.
Opensips versions I tried: 3.4.4-1 and 3.4.3-1.

I make a call from a phone behind nat to sip server via opensips, so I 
use the fix_contact function from nat_traversal/fix_nated_contact from 
nathelper for each request.
On the first invite I get "caller_contact": 
"sip:1002@3.3.3.1:49682;transport=TCP;ob" in "opensips-cli -x mi 
dlg_list".
If caller side makes re-invite, then i see this "caller_contact": 
"\u0001" for nat_traversal,
and this "caller_contact": "sip:1002@3.3.3.1 
<mailto:sip%3A1002@3.3.3.1>\ufffd" with nathelper in dlg_list.
In that case dlg_send_sequential mi dialog function cannot be used 
cause caller contact cannot be read.


Also if i make a re-invite from the caller side then the 
caller_contact field in dlg_list starts looking normal.


Am I doing something wrong with fixing the nated contact field?


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Re: [OpenSIPS-Users] infinite loop in rtpengine rtpe_function_call()

2024-02-23 Thread Bogdan-Andrei Iancu

Hi Alexander,

Thank for your report - it will be better to push this via the github 
tracker, the right place for coding issues:

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

Bogdan-Andrei Iancu

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

On 22.02.2024 21:43, Alexander Kogan wrote:


Hi,

The OpenSIPS process gets stack in infinite loop when the rtpengine 
module runs out of memory in send_rtpe_command()


Please review the following piece of code in rtpengine.c:

    RTPE_START_READ();
    do {
    if (snode && snode->s) {
    if ((node = get_rtpe_node(snode, set)) == NULL && op == 
OP_OFFER)

    node = select_rtpe_node(ng_flags.call_id, 1, set);
    snode = NULL;
    } else {
    node = select_rtpe_node(ng_flags.call_id, 1, set);
    }
    if (!node) {
    LM_ERR("no available proxies\n");
    RTPE_STOP_READ();
    goto error;
    }

    cp = send_rtpe_command(node, ng_flags.dict, );
    } *while (cp == NULL);*
    RTPE_STOP_READ();
    LM_DBG("proxy reply: %.*s\n", ret, cp);

When send_rtpe_command() returns NULL due to memory absence here:

    v = bencode_iovec(dict, , 1, 0);
    if (!v) {
    LM_ERR("error converting bencode to iovec\n");
    return NULL;
    }

the mentioned loop ( *do {} while(cp == NULL)* ) won't stop 
indefinitely causing the opensips process to become stuck.


Best regards,
Alexander Kogan,
Director of R
5g Future
http://5gfuture.com

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