[OpenSIPS-Users] OpenSIPS 3.5.0 goes stable

[Bogdan-Andrei Iancu]

 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 

*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



Do you want to learn more on OpenSIPS 3.5?

Join us for the firsts 3.5 *OpenSIPS eBootcamp training* 
, 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 !




--
Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] [OpenSIPS-Devel] OpenSIPS 3.5.0 major release, beta version

[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 
 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 
 or the GIT repo 
.


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

[Bogdan-Andrei Iancu]

 14th - 25th October 2024,


 online, worldwide



*Study smarter, not harder!
*

Take advantage of the *OpenSIPS Bootcamp* 
 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 



*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 



*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 !


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 
.



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  Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
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Re: [OpenSIPS-Users] Opensips as proxy for Asterisk

[sterlin]

  
  
  
Hi,   
  
I am facing issue with INVITE .
  

  

  

  
  
  
  
  
>   
> On 31 May 2024 at 3:46 pm, Bogdan-Andrei Iancuwrote:
>   
>   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
> >   
> >
> >   
> > 
> >  ___ Users mailing list  
> > Users@lists.opensips.org   
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users   
>
>   
>   
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Re: [OpenSIPS-Users] Opensips as proxy for Asterisk

[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] opensips nat question

[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] Opensips as proxy for Asterisk

[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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[OpenSIPS-Users] OpenSIPS Summit 2024 - post facts

[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
   
 * the recordings are also available on the OpenSIPS YouTube channel
   
 * photos from the event were uploaded on the OpenSIPS Summit 2024
   Album 


Enjoy !

--
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] opensips not failing over on 500.

[Ben Newlin]

Yes, the default behavior of failure_route is to revert the received reply back 
upstream [1]. If failover is desired a new branch must be created by some 
action in the failure_route. I don’t use load_balancer but I believe for that 
module this would be the lb_next function [2].

[1] - https://www.opensips.org/Documentation/Script-Routes-3-5#toc3
[2] - https://opensips.org/docs/modules/3.5.x/load_balancer.html#func_lb_next

Ben Newlin

From: Users  on behalf of Johan De Clercq 

Date: Wednesday, May 29, 2024 at 3:56 AM
To: OpenSIPS users mailling list 
Subject: [OpenSIPS-Users] opensips not failing over on 500.
 EXTERNAL EMAIL - Please use caution with links and attachments


Hi,

I think that this is by design.
when you use load_balancer and you want to failover on receiving certain cause 
codes, then you need to do that in failure_route, I believe.  Opensips will nto 
failover automatically to the next destination when receiving 500.

Am I correct with my assumption ?

Br, Johan.
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[OpenSIPS-Users] opensips not failing over on 500.

[Johan De Clercq]

Hi,

I think that this is by design.
when you use load_balancer and you want to failover on receiving certain
cause codes, then you need to do that in failure_route, I believe.
Opensips will nto failover automatically to the next destination when
receiving 500.

Am I correct with my assumption ?

Br, Johan.
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[OpenSIPS-Users] OpenSIPS 3.5.0 major release, beta version

[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 
 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 
 or the GIT repo 
.


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] opensips call center module

[Prathibha B]

Exactly after 31 seconds, the video becomes blank. What could be the reason?

*Console Log:*
Wed May 08 2024 09:11:03 GMT+0530 (India Standard Time) | sip.invite-dialog
| No ACK for 2xx response received, attempting retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:05 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:09 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:13 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:17 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:21 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:25 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:29 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.invite-dialog | No ACK for 2xx response received, attempting
retransmission
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.Invitation | Invitation.onAckTimeout
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.Invitation | No ACK received for an extended period of time,
terminating session
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.invite-dialog | INVITE dialog
B2B.245.4610557.1715138504.19934727797hqeu982nm5b311b272f7dc2a4783f192166366d5a-4bdb
sending BYE request
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.invite-dialog | INVITE dialog
B2B.245.4610557.1715138504.19934727797hqeu982nm5b311b272f7dc2a4783f192166366d5a-4bdb
destroyed
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.Invitation | Session
B2B.245.4610557.1715138504.19934727795b311b272f7dc2a4783f192166366d5a-4bdb
transitioned to state Terminated
sip-0.20.0.min.js:2 Wed May 08 2024 09:11:33 GMT+0530 (India Standard Time)
| sip.Invitation | Session
B2B.245.4610557.1715138504.19934727795b311b272f7dc2a4783f192166366d5a-4bdb
in state Terminated is being disposed

On Tue, 7 May 2024 at 17:20, Prathibha B  wrote:

> Getting blank screen after 32 seconds.
>
> On Tue, 7 May 2024 at 13:31, Prathibha B  wrote:
>
>> Yes. It always stay in incall state.
>>
>> On Tue, 7 May 2024 at 13:14, Jehanzaib Younis 
>> wrote:
>>
>>> Does it always stay in the  incall State? If this is the case then you
>>> need to check somehow BYE or CANCEL not working properly.
>>>
>>> Regards,
>>> Jehanzaib
>>>
>>>
>>> On Tue, May 7, 2024 at 6:03 PM Prathibha B 
>>> wrote:
>>>
 "Agents": [
 {
 "id": "101001",
 "Ref": 1,
 "Loged in": "YES",
 "State": "incall"
 },
 {
 "id": "101002",
 "Ref": 1,
 "Loged in": "YES",
 "State": "incall"
 }
 ]

 The agent state shows status as incall.

 On Tue, 7 May 2024 at 06:35, Jehanzaib Younis <
 jehanzaib.ki...@gmail.com> wrote:

> Hi Parathiba,
>
> Could you capture the SIP packets? They'll provide insight into what's
> going on.
>
>
> Regards,
> Jehanzaib
>
>
> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
> wrote:
>
>> I'm able to hear the message given in the message queue but the call
>> is not getting transferred to the online agent,
>>
>> --
>> Regards,
>> B.Prathibha
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>


 --
 Regards,
 B.Prathibha
 ___
 Users mailing list
 Users@lists.opensips.org
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>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
Regards,
B.Prathibha
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Re: [OpenSIPS-Users] opensips call center module

[Prathibha B]

Yes. It always stay in incall state.

On Tue, 7 May 2024 at 13:14, Jehanzaib Younis 
wrote:

> Does it always stay in the  incall State? If this is the case then you
> need to check somehow BYE or CANCEL not working properly.
>
> Regards,
> Jehanzaib
>
>
> On Tue, May 7, 2024 at 6:03 PM Prathibha B 
> wrote:
>
>> "Agents": [
>> {
>> "id": "101001",
>> "Ref": 1,
>> "Loged in": "YES",
>> "State": "incall"
>> },
>> {
>> "id": "101002",
>> "Ref": 1,
>> "Loged in": "YES",
>> "State": "incall"
>> }
>> ]
>>
>> The agent state shows status as incall.
>>
>> On Tue, 7 May 2024 at 06:35, Jehanzaib Younis 
>> wrote:
>>
>>> Hi Parathiba,
>>>
>>> Could you capture the SIP packets? They'll provide insight into what's
>>> going on.
>>>
>>>
>>> Regards,
>>> Jehanzaib
>>>
>>>
>>> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
>>> wrote:
>>>
 I'm able to hear the message given in the message queue but the call is
 not getting transferred to the online agent,

 --
 Regards,
 B.Prathibha
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>


-- 
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B.Prathibha
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Re: [OpenSIPS-Users] opensips call center module

[Jehanzaib Younis]

Does it always stay in the  incall State? If this is the case then you need
to check somehow BYE or CANCEL not working properly.

Regards,
Jehanzaib


On Tue, May 7, 2024 at 6:03 PM Prathibha B  wrote:

> "Agents": [
> {
> "id": "101001",
> "Ref": 1,
> "Loged in": "YES",
> "State": "incall"
> },
> {
> "id": "101002",
> "Ref": 1,
> "Loged in": "YES",
> "State": "incall"
> }
> ]
>
> The agent state shows status as incall.
>
> On Tue, 7 May 2024 at 06:35, Jehanzaib Younis 
> wrote:
>
>> Hi Parathiba,
>>
>> Could you capture the SIP packets? They'll provide insight into what's
>> going on.
>>
>>
>> Regards,
>> Jehanzaib
>>
>>
>> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
>> wrote:
>>
>>> I'm able to hear the message given in the message queue but the call is
>>> not getting transferred to the online agent,
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
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>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Regards,
> B.Prathibha
> ___
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>
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Re: [OpenSIPS-Users] opensips call center module

[Prathibha B]

After about 45 seconds, the screen on the receiver side becomes blank.

On Tue, 7 May 2024 at 12:32, Prathibha B  wrote:

> I delete the records from cc_calls table and restart opensips. Then the
> call is getting is transferred to one of the agents. What is the solution
> to this? Everytime it is not possible to delete the records in cc_calls
> table and restart opensips.
>
> When I reject the call, the call is transferred to the same agent. The
> call is not going to the next available agent.
>
> On Tue, 7 May 2024 at 11:32, Prathibha B  wrote:
>
>> "Agents": [
>> {
>> "id": "101001",
>> "Ref": 1,
>> "Loged in": "YES",
>> "State": "incall"
>> },
>> {
>> "id": "101002",
>> "Ref": 1,
>> "Loged in": "YES",
>> "State": "incall"
>> }
>> ]
>>
>> The agent state shows status as incall.
>>
>> On Tue, 7 May 2024 at 06:35, Jehanzaib Younis 
>> wrote:
>>
>>> Hi Parathiba,
>>>
>>> Could you capture the SIP packets? They'll provide insight into what's
>>> going on.
>>>
>>>
>>> Regards,
>>> Jehanzaib
>>>
>>>
>>> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
>>> wrote:
>>>
 I'm able to hear the message given in the message queue but the call is
 not getting transferred to the online agent,

 --
 Regards,
 B.Prathibha
 ___
 Users mailing list
 Users@lists.opensips.org
 http://lists.opensips.org/cgi-bin/mailman/listinfo/users

>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>
>>
>> --
>> Regards,
>> B.Prathibha
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
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B.Prathibha
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Re: [OpenSIPS-Users] opensips call center module

[Prathibha B]

I delete the records from cc_calls table and restart opensips. Then the
call is getting is transferred to one of the agents. What is the solution
to this? Everytime it is not possible to delete the records in cc_calls
table and restart opensips.

When I reject the call, the call is transferred to the same agent. The call
is not going to the next available agent.

On Tue, 7 May 2024 at 11:32, Prathibha B  wrote:

> "Agents": [
> {
> "id": "101001",
> "Ref": 1,
> "Loged in": "YES",
> "State": "incall"
> },
> {
> "id": "101002",
> "Ref": 1,
> "Loged in": "YES",
> "State": "incall"
> }
> ]
>
> The agent state shows status as incall.
>
> On Tue, 7 May 2024 at 06:35, Jehanzaib Younis 
> wrote:
>
>> Hi Parathiba,
>>
>> Could you capture the SIP packets? They'll provide insight into what's
>> going on.
>>
>>
>> Regards,
>> Jehanzaib
>>
>>
>> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
>> wrote:
>>
>>> I'm able to hear the message given in the message queue but the call is
>>> not getting transferred to the online agent,
>>>
>>> --
>>> Regards,
>>> B.Prathibha
>>> ___
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Regards,
> B.Prathibha
>


-- 
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B.Prathibha
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Re: [OpenSIPS-Users] opensips call center module

[Prathibha B]

"Agents": [
{
"id": "101001",
"Ref": 1,
"Loged in": "YES",
"State": "incall"
},
{
"id": "101002",
"Ref": 1,
"Loged in": "YES",
"State": "incall"
}
]

The agent state shows status as incall.

On Tue, 7 May 2024 at 06:35, Jehanzaib Younis 
wrote:

> Hi Parathiba,
>
> Could you capture the SIP packets? They'll provide insight into what's
> going on.
>
>
> Regards,
> Jehanzaib
>
>
> On Tue, May 7, 2024 at 12:40 AM Prathibha B 
> wrote:
>
>> I'm able to hear the message given in the message queue but the call is
>> not getting transferred to the online agent,
>>
>> --
>> Regards,
>> B.Prathibha
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
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B.Prathibha
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Re: [OpenSIPS-Users] opensips call center module

[Jehanzaib Younis]

Hi Parathiba,

Could you capture the SIP packets? They'll provide insight into what's
going on.


Regards,
Jehanzaib


On Tue, May 7, 2024 at 12:40 AM Prathibha B 
wrote:

> I'm able to hear the message given in the message queue but the call is
> not getting transferred to the online agent,
>
> --
> Regards,
> B.Prathibha
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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[OpenSIPS-Users] opensips call center module

[Prathibha B]

I'm able to hear the message given in the message queue but the call is not
getting transferred to the online agent,

-- 
Regards,
B.Prathibha
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[OpenSIPS-Users] opensips

[Prathibha B]

Does opensips call center module work for webrtc based video calls?

-- 
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B.Prathibha
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[OpenSIPS-Users] OpenSIPS Summit 2024 - This is our content

[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



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 

**









--
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  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue

[Luis Leal via Users]

Hi,

I've done some exploring of the source code and have a hunch as to what's 
happening.

The full trace is as follows where:
1.  Endpoints = 10.1.17.12 / 10.249.224.9
2.  SRS = 10.1.17.24
3.  OpenSIPS = 10.1.17.15

12  21:33:50.208348 10.1.17.15  506010.249.224.954204   SIP/SDP 
1125Request: INVITE sip:1125@10.249.224.9:54204;ob |
13  21:33:50.235594 10.249.224.954204   10.1.17.15  5060SIP 
539 Status: 100 Trying |
14  21:33:50.235938 10.249.224.954204   10.1.17.15  5060SIP 
725 Status: 180 Ringing |
15  21:33:50.236090 10.1.17.15  506010.1.17.12  5060SIP 
641 Status: 180 Ringing |
20  21:33:53.262006 10.249.224.954204   10.1.17.15  5060SIP/SDP 
1171Status: 200 OK (INVITE) |
21  21:33:53.262560 10.1.17.15  506010.1.17.12  5060SIP/SDP 
1073Status: 200 OK (INVITE) |
22  21:33:53.263170 10.1.17.12  506010.1.17.15  5060SIP 
532 Request: ACK sip:1125@10.249.224.9:54204;ob |
24  21:33:53.263325 10.1.17.15  506010.1.17.24  5060
SIP/SDP/XML 1026Request: INVITE sip:10.1.17.24:5060 |

The errors logged:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

It looks like either the callback tm_start_recording is being called too early 
on the 1XX packets or rtp_relay_copy_offer isn’t handling the unconfirmed 
session correctly?

And perhaps the invite to the SRS shouldn’t be going out if there’s no RTP 
stream?

I’m of course not an expert in the OpenSIPS architecture so I could be wrong. :)

I’d appreciate it if someone more knowledgeable could confirm.

Kind regards

Luis

Date: Tue, 23 Apr 2024 08:39:33 +
From: Luis Leal mailto:lu...@scarab.co.za>>
To: "users@lists.opensips.org<mailto:users@lists.opensips.org>" 
mailto:users@lists.opensips.org>>
Subject: [OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue
Message-ID: 
<47660a2ed6764bc58081e8e2a54f1...@scarab.co.za<mailto:47660a2ed6764bc58081e8e2a54f1...@scarab.co.za>>
Content-Type: text/plain; charset="utf-8"

Hi there,

We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 
and I was hoping someone would be able to shed some light on it.

There are two symptoms:

  1.  Errors in the opensips log
  2.  SIPREC invite with correct SDP details (as per rtpengine log) but stream 
metadata missing from the XML metadata

The errors in the log are as follows:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

The curious part is that the above error happens before the 200 OK is received. 
The relevant SIP trace is:

12   21:33:50.20834810.1.17.15 5060 
10.249.224.954204   SIP/SDP   1125 Request: INVITE 
sip:1125@10.249.224.9:54204;ob |

...Snip...

21   21:33:53.26256010.1.17.15 5060 10.1.17.12  
   5060 SIP/SDP   1073 Status: 200 OK (INVITE) |

The SIPREC invite is still generated though but is missing stream details 
(participant details masked with +27X for privacy):

Session Initiation Protocol (SIP as raw text)
INVITE sip:10.1.17.24:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0
To: sip:10.1.17.24:5060
From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21
CSeq: 2 INVITE
Call-ID: B2B.364.22430.1713814432.535273629
Max-Forwards: 70
Content-Length: 1995
User-Agent: OpenSIPS (3.4.4 (x86_64/linux))
Require: siprec
Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42
Contact: sip:10.1.17.15:5060;+sip.src

--OSS-unique-boundary-42
Content-Type: application/sdp

v=0
o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8
s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9
t=0 0
m=audio 31432 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1120210035 cname:060168be20ab122b
a=sendonly
a=rtcp:31433
m=audio 36760 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
a=rtcp:36761
a=ptime:20

--OSS-unique-boundary-42
Content-Type: application/rs-metad

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

[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


Orange Restricted


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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 
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[OpenSIPS-Users] OpenSIPS 3.4 SIPREC Issue

[Luis Leal via Users]

Hi there,

We're encountering a curious issue with SIPREC in upgrading from 3.2 to 3.4.4 
and I was hoping someone would be able to shed some light on it.

There are two symptoms:

  1.  Errors in the opensips log
  2.  SIPREC invite with correct SDP details (as per rtpengine log) but stream 
metadata missing from the XML metadata

The errors in the log are as follows:

Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:rtp_relay:rtp_relay_copy_offer: rtp not established!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:src_start_recording: could not start recording!
Apr 22 21:33:50 vltelcceprd211 /usr/sbin/opensips[902536]: 
ERROR:siprec:tm_start_recording: cannot start recording!

The curious part is that the above error happens before the 200 OK is received. 
The relevant SIP trace is:

12   21:33:50.20834810.1.17.15 5060 
10.249.224.954204   SIP/SDP   1125 Request: INVITE 
sip:1125@10.249.224.9:54204;ob |

...Snip...

21   21:33:53.26256010.1.17.15 5060 10.1.17.12  
   5060 SIP/SDP   1073 Status: 200 OK (INVITE) |

The SIPREC invite is still generated though but is missing stream details 
(participant details masked with +27X for privacy):

Session Initiation Protocol (SIP as raw text)
INVITE sip:10.1.17.24:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.17.15:5060;branch=z9hG4bK0235.aca30813.0
To: sip:10.1.17.24:5060
From: sip:10.1.17.24:5060;tag=c5d35275eae8a009626d3007dc8441a2-ce21
CSeq: 2 INVITE
Call-ID: B2B.364.22430.1713814432.535273629
Max-Forwards: 70
Content-Length: 1995
User-Agent: OpenSIPS (3.4.4 (x86_64/linux))
Require: siprec
Content-Type: multipart/mixed;boundary=OSS-unique-boundary-42
Contact: sip:10.1.17.15:5060;+sip.src

--OSS-unique-boundary-42
Content-Type: application/sdp

v=0
o=- 7360776941148045834 7360776941148045834 IN IP4 10.1.17.8
s=rtpengine-12-3-1-2-0-mr12-3-1-2-1-el9
t=0 0
m=audio 31432 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:0
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ssrc:1120210035 cname:060168be20ab122b
a=sendonly
a=rtcp:31433
m=audio 36760 RTP/AVP 8 101
c=IN IP4 10.1.17.8
a=label:1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendonly
a=rtcp:36761
a=ptime:20

--OSS-unique-boundary-42
Content-Type: application/rs-metadata+xml
Content-Disposition: recording-session



 complete
 
  2d409cb6-b066-4579-8c49-c6e6a7b9d600
 
 
  
   +27X
  
 
 
  
 
 
  2024-04-22T21:33:50+0200
 
 
  2024-04-22T21:33:50+0200
 
 
  2024-04-22T21:33:50+0200
 
 
 
 
 

--OSS-unique-boundary-42--

Is there a configuration item we're missing perhaps?

Kind regards

Luis Leal



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[OpenSIPS-Users] OpenSIPS 3.5 release dates

[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,

--
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OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
  https://www.siphub.com
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Re: [OpenSIPS-Users] openSIPS with an AS that would handle the routing logic

[Vlad Paiu]

Hello,

Since OpenSIPS is a SIP server, I would try to leverage it's 
capabilities and communicate over SIP with the AS - that means scripting 
your way through.



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


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 
modified, changed or falsified.
Thank you.

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

[julien . royannais]

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



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 
modified, changed or falsified.
Thank you.
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[OpenSIPS-Users] Opensips as proxy for Asterisk

[Sterlin Devanish]

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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[OpenSIPS-Users] OpenSIPS Summit 2024 - Speaker's lineup

[Bogdan-Andrei Iancu]

 OpenSIPS Summit


 May 14th - 17th, 2024

Valencia, Spain


*Speaker's lineup & Schedule
*

We bring here the list of speakers and papers 
 - great 
speakers presenting great topics to share experience and knowledge to a 
great audience. Explore here all the details...




*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 

**









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

[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  ?

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

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

[Sasmita Panda]

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  ?

*Thanks & Regards*
*Sasmita Panda*
*Senior Network Testing and Software Engineer*
*3CLogic , ph:07827611765*
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK [SOLVED]

[Social Boh]

Right, solved

Thank you

---
I'm SoCIaL, MayBe

El 20/03/2024 a las 4:07 a. m., Ihor Olkhovskyi escribió:

You are not increasing CSeq for subsequent PRACK.

Le mar. 19 mars 2024 à 18:58, Social Boh  a écrit :

Some progress

Now I can reply the first 183 with the PRACK response but when the
provider send a 180 ringing and I reply with another PPRACK
response the provider reply with 500 Server Internal Error

Annex PCAP Capture.

Any hint is really appreciate.

Regards

---
I'm SoCIaL, MayBe

El 15/03/2024 a las 5:35 p. m., Social Boh escribió:


I'm trying using directly opensips-cli via exec module to see if
all OK and then pass the data tu mi_script module

I think I have a sintaxis problem with the line:

exec("opensips-cli -x mi t_uac_dlg method=INVITE
ruri="sip:alice@127.0.0.1:7050" headers="From:
sip:bobster@127.0.0.1:1337\r\nTo:
sip:alice@127.0.0.1:7050\r\nContact:
sip:bobster@127.0.0.1:1337\r\n"")

I don't know how use " in the line because Headers and ruri need
them. The result is a error:

ERROR:core:handle_mi_request: Invalid parameters

Any hint, please?

Regards

---
I'm SoCIaL, MayBe
El 26/02/2024 a las 7:43 a. m., Bogdan-Andrei Iancu escribió:

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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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-- 
Best regards,

Ihor (Igor)

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--
Best 

[OpenSIPS-Users] Opensips as frontend proxy server and Asterisk as backend registrar server

[Sterlin Devanish]

Hi all,
 Is it possible to configure opensips as the frontend proxy server and
Asterisk as the backend registrar server to handle RFC8599 for the Flutter
WebRTC client?

*Thanks & Regards,*
*Sterlin Devanish D*
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Ihor Olkhovskyi]

You are not increasing CSeq for subsequent PRACK.

Le mar. 19 mars 2024 à 18:58, Social Boh  a écrit :

> Some progress
>
> Now I can reply the first 183 with the PRACK response but when the
> provider send a 180 ringing and I reply with another PPRACK response the
> provider reply with 500 Server Internal Error
>
> Annex PCAP Capture.
>
> Any hint is really appreciate.
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
> El 15/03/2024 a las 5:35 p. m., Social Boh escribió:
>
> I'm trying using directly opensips-cli via exec module to see if all OK
> and then pass the data tu mi_script module
>
> I think I have a sintaxis problem with the line:
>
> exec("opensips-cli -x mi t_uac_dlg method=INVITE ruri=
> "sip:alice@127.0.0.1:7050" headers="From: sip:bobster@127.0.0.1:1337\r\nTo:
> sip:alice@127.0.0.1:7050\r\nContact: sip:bobster@127.0.0.1:1337\r\n"")
>
> I don't know how use " in the line because Headers and ruri need them. The
> result is a error:
>
> ERROR:core:handle_mi_request: Invalid parameters
>
> Any hint, please?
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
> El 26/02/2024 a las 7:43 a. m., Bogdan-Andrei Iancu escribió:
>
> 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
>> >
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Best regards,
> Ihor (Igor)
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


-- 
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Ihor (Igor)
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

Some progress

Now I can reply the first 183 with the PRACK response but when the 
provider send a 180 ringing and I reply with another PPRACK response the 
provider reply with 500 Server Internal Error


Annex PCAP Capture.

Any hint is really appreciate.

Regards

---
I'm SoCIaL, MayBe

El 15/03/2024 a las 5:35 p. m., Social Boh escribió:


I'm trying using directly opensips-cli via exec module to see if all 
OK and then pass the data tu mi_script module


I think I have a sintaxis problem with the line:

exec("opensips-cli -x mi t_uac_dlg method=INVITE 
ruri="sip:alice@127.0.0.1:7050" headers="From: 
sip:bobster@127.0.0.1:1337\r\nTo: sip:alice@127.0.0.1:7050\r\nContact: 
sip:bobster@127.0.0.1:1337\r\n"")


I don't know how use " in the line because Headers and ruri need them. 
The result is a error:


ERROR:core:handle_mi_request: Invalid parameters

Any hint, please?

Regards

---
I'm SoCIaL, MayBe
El 26/02/2024 a las 7:43 a. m., Bogdan-Andrei Iancu escribió:
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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
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--
Best regards,
Ihor (Igor)

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PRACK.pcap
Description: Binary data
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

I'm trying using directly opensips-cli via exec module to see if all OK 
and then pass the data tu mi_script module


I think I have a sintaxis problem with the line:

exec("opensips-cli -x mi t_uac_dlg method=INVITE 
ruri="sip:alice@127.0.0.1:7050" headers="From: 
sip:bobster@127.0.0.1:1337\r\nTo: sip:alice@127.0.0.1:7050\r\nContact: 
sip:bobster@127.0.0.1:1337\r\n"")


I don't know how use " in the line because Headers and ruri need them. 
The result is a error:


ERROR:core:handle_mi_request: Invalid parameters

Any hint, please?

Regards

---
I'm SoCIaL, MayBe

El 26/02/2024 a las 7:43 a. m., Bogdan-Andrei Iancu escribió:
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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
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--
Best regards,
Ihor (Igor)

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

Hello,

two questions about mi_script module:

1. I don't understand well this sentence: "Moreover, if the running MI
   /command / is configured to run in asynchronous mode (such as
   /t_uac_dlg/ the command blocks in a busy waiting manner until the
   response is received." can I or I can't use t_uac_dlg in
   asynchronous mode?
2. Can I indicate more than un parameters in the line like:
 * $avp(params) = "method","callid","From:";

 * or is one value for line? Same for $avp(vals)

Thank you

Regards

---
I'm SoCIaL, MayBe

El 26/02/2024 a las 7:43 a. m., Bogdan-Andrei Iancu escribió:
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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
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--
Best regards,
Ihor (Igor)

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

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Re: [OpenSIPS-Users] opensips 3.4 push notification

[Bogdan-Andrei Iancu]

Check with 
https://blog.opensips.org/2020/06/03/sip-push-notification-with-opensips-3-1-lts-rfc-8599-supportpart-ii/


If setting $var(do_relay) to "false", be sure you do not hit any 
t_relay() later in your script.


Regards,

Bogdan-Andrei Iancu

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

On 20.02.2024 17:44, r...@rvgeerligs.nl wrote:

As far as I can see I do not do a relay in case 2:

lookup("location", "method-filtering");
$var(rc) = $retcode;
switch ($var(rc)) {
case 1:
    # we found at least 1 non-PN contact!
    $var(do_relay) = true;
    break;
case 2:
    # success, but all contacts are PN-enabled, so we're
    # sending PNs / awaiting re-registrations from them
*    $var(do_relay) = false;*
#     $var(do_relay) = true;

    break;
default:
    xlog("L_INFO", "DBG: no contacts found ($var(rc))\n");
    t_reply(404, "Not Found");
    exit;
}

Please advise again,

Regards,

Ronald


February 19, 2024 at 6:57 AM, "Bogdan-Andrei Iancu" 
> 
wrote:


Hi Ronald,

Do you check in the cfg the return code of the lookup() [1]
function for retcode (2)? you should not perform any SIP relay in
this case, as it will result into a loop.


[1]
https://opensips.org/html/docs/modules/3.4.x/registrar.html#func_lookup

Regards,

Bogdan-Andrei Iancu

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

On 16.02.2024 18:44, r...@rvgeerligs.nl wrote:

here you are.

February 16, 2024 at 1:37 PM, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org?to=%22Bogdan-Andrei%20Iancu%22%20%3Cbogdan%40opensips.org%3E>>
wrote:

Hi,

Could you share a pcap or trace so we can understand the
routing you have ?

Regards,

Bogdan-Andrei Iancu

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

On 15.02.2024 22:24, r...@rvgeerligs.nl wrote:

Hi,
I kind of followed the article SIP Push Notification
with OpenSIPS 3.1 LTS [RFC 8599 support][Part II] as I
connot find anything related to opensips 3.4.x.
When I disable (commenting by #) modparam("registrar",
"pn_enable", true), the call actually works.
When I enable modparam("registrar", "pn_enable", true)
the call does not come through.
It gets rerouted internally to external ip address
after the authentication, second INVITE with answer to
nonce, and giving it a try.
Opensips keeps asking for Proxy Authentication
required which was allready accepted.
Anyone an idea?
Regards,
Ronald Geerligs

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Re: [OpenSIPS-Users] opensips 3.4 push notification

[Bogdan-Andrei Iancu]

Hi Ronald,

Do you check in the cfg the return code of the lookup() [1] function for 
retcode (2)? you should not perform any SIP relay in this case, as it 
will result into a loop.



[1] https://opensips.org/html/docs/modules/3.4.x/registrar.html#func_lookup

Regards,

Bogdan-Andrei Iancu

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

On 16.02.2024 18:44, r...@rvgeerligs.nl wrote:

here you are.

February 16, 2024 at 1:37 PM, "Bogdan-Andrei Iancu" 
> 
wrote:


Hi,

Could you share a pcap or trace so we can understand the routing
you have ?

Regards,

Bogdan-Andrei Iancu

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

On 15.02.2024 22:24, r...@rvgeerligs.nl wrote:

Hi,
I kind of followed the article SIP Push Notification with
OpenSIPS 3.1 LTS [RFC 8599 support][Part II] as I connot find
anything related to opensips 3.4.x.
When I disable (commenting by #) modparam("registrar",
"pn_enable", true), the call actually works.
When I enable modparam("registrar", "pn_enable", true) the
call does not come through.
It gets rerouted internally to external ip address after the
authentication, second INVITE with answer to nonce, and giving
it a try.
Opensips keeps asking for Proxy Authentication required which
was allready accepted.
Anyone an idea?
Regards,
Ronald Geerligs

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Re: [OpenSIPS-Users] opensips 3.4 push notification

[Bogdan-Andrei Iancu]

Hi,

Could you share a pcap or trace so we can understand the routing you have ?

Regards,

Bogdan-Andrei Iancu

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

On 15.02.2024 22:24, r...@rvgeerligs.nl wrote:

Hi,

I kind of followed the article SIP Push Notification with OpenSIPS 3.1 
LTS [RFC 8599 support][Part II] as I connot find anything related to 
opensips 3.4.x.


When I disable (commenting by #) modparam("registrar", "pn_enable", 
true), the call actually works.
When I enable modparam("registrar", "pn_enable", true) the call does 
not come through.


It gets rerouted internally to external ip address after the 
authentication, second INVITE with answer to nonce, and giving it a try.
Opensips keeps asking for Proxy Authentication required which was 
allready accepted.



Anyone an idea?

Regards,


Ronald Geerligs

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Bogdan-Andrei Iancu]

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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
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--
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Ihor (Igor)

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[OpenSIPS-Users] opensips 3.4 push notification

[rvg]

Hi,

I kind of followed the article SIP Push Notification with OpenSIPS 3.1 LTS [RFC 
8599 support][Part II] as I connot find anything related to opensips 3.4.x.

When I disable (commenting by #) modparam("registrar", "pn_enable", true), the 
call actually works.
When I enable modparam("registrar", "pn_enable", true) the call does not come 
through.

It gets rerouted internally to external ip address after the authentication, 
second INVITE with answer to nonce, and giving it a try.
Opensips keeps asking for Proxy Authentication required which was allready 
accepted.


Anyone an idea?

Regards,


Ronald Geerligs___
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Re: [OpenSIPS-Users] opensips database

[Alexey]

Hello,

isn't it what you need?
https://www.opensips.org/Documentation/TipsFAQ#toc7


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https://alexeyka.zantsev.com/

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[OpenSIPS-Users] opensips database

[Prathibha B]

How to use encrypted password in opensips?

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[M S]

After 302, when you receive 183 from the provider (that expects PRACK), you
forward that 183 to the other side of conversation - and that side is
responsible for generating PRACK and sending it to opensips to forward to
the provider.

On Mon, Feb 12, 2024 at 1:16 PM Social Boh  wrote:

> Hello,
>
> the flow chart is this:
>
> -> INVITE (to OpensiSIP)
>
> Relay to Redirect Server
>
> Redirect server reply with a 302 and a contact that OpenSIPs use to create
>
> INVITE (OpenSIPs) ->
>
> <- 183 (provider) with  Require: 100rel
>
> PRACK (from OpenSIPs)
>
> I don't know how create this PRACK
>
> I have:
>
> modparam("b2b_entities", "passthru_prack", 0)
>
> but this is not enough and I have to use (I think)
>
> b2b_init_request("top hiding");
>
> but this function works only on Request Route and I catch the 302 in the
> failure route.
>
> Thank you
>
> Regards
>
> ---
> I'm SoCIaL, MayBe
>
> El 12/02/2024 a las 4:36 a. m., Ihor Olkhovskyi escribió:
>
> 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
>> >
>> >
>> > ___
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
>
> --
> Best regards,
> Ihor (Igor)
>
> ___
> Users mailing 
> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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>
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

Hello,

the flow chart is this:

-> INVITE (to OpensiSIP)

Relay to Redirect Server

Redirect server reply with a 302 and a contact that OpenSIPs use to create

INVITE (OpenSIPs) ->

<- 183 (provider) with  Require: 100rel

PRACK (from OpenSIPs)

I don't know how create this PRACK

I have:

modparam("b2b_entities", "passthru_prack", 0)

but this is not enough and I have to use (I think)

b2b_init_request("top hiding");

but this function works only on Request Route and I catch the 302 in the 
failure route.


Thank you

Regards

---
I'm SoCIaL, MayBe

El 12/02/2024 a las 4:36 a. m., Ihor Olkhovskyi escribió:
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
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users

___
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--
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Ihor (Igor)

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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Ihor Olkhovskyi]

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
> >
> >
> > ___
> > Users mailing list
> > Users@lists.opensips.org
> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>


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Ihor (Igor)
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Re: [OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

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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[OpenSIPS-Users] OpenSIPs generate PRACK

[Social Boh]

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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Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

[Alexey]

excuse me, posted message to wrong thread

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Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

[Alexey]

Is it possible to cache not exact number of table row
but all values from concrete column from all rows?

The total amount of rows in table is not big, several rows or several dozens.

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Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

[Prathibha B]

I want to setup a call center.

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

From: Users  on behalf of Alexey 

Sent: Friday, February 9, 2024 12:24:11 AM
To: OpenSIPS users mailling list 
Subject: Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

Hello,

What exact kind of integration do you mean?
Any peculiarities of the 18.20 release?

There is a tutorial [1] but for OpenSIPS 1.8 and Asterisk 1.8.
So, the main difference according to Asterisk is moving
from chan_sip towards chan_pjsip.

[1] 
https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8


--
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https://alexeyka.zantsev.com/

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Re: [OpenSIPS-Users] OpenSIPS - Asterisk Integration

[Alexey]

Hello,

What exact kind of integration do you mean?
Any peculiarities of the 18.20 release?

There is a tutorial [1] but for OpenSIPS 1.8 and Asterisk 1.8.
So, the main difference according to Asterisk is moving
from chan_sip towards chan_pjsip.

[1] 
https://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration-1-8


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https://alexeyka.zantsev.com/

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[OpenSIPS-Users] OpenSIPS - Asterisk Integration

[Prathibha B]

Has anyone done the integration of Opensips 3.4 with Asterisk 18.20?

-- 
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B.Prathibha
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[OpenSIPS-Users] OpenSIPS IMS at Fosdem'24

[Răzvan Crainea]

Hi, Everyone!

This year history repeats itself, thus myself (Răzvan Crainea) and Liviu 
Chircu will be representing OpenSIPS at the Fosdem'24 conference, where 
will be talking about how you can Provide VoLTE and/or VoNR for IMS 
using OpenSIPS 3.5.
This talk will present the way OpenSIPS is tackling its IMS 
implementation, and will contain a lot of the topics that we developed
within the OpenSIPS IMS Working Group[1] - so if you are interested in 
the topic, make sure you subscribe (if you haven't already) to the group 
and bring your valuable contribution the the IMS topic.
Our presentation starts on Saturday, 17:50[1] in room H.1302, Real Time 
Communications (RTC) devroom[2]. We hope to see as many of you as possible!


[1] http://lists.opensips.org/cgi-bin/mailman/listinfo/wg-ims
[2] 
https://fosdem.org/2024/schedule/event/fosdem-2024-3614-provide-volte-vonr-using-opensips-3-5/


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

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Re: [OpenSIPS-Users] opensips 3.4 documentation event_routing example 1.6.1 - push notification

[Bogdan-Andrei Iancu]

Hi Ronald,

Thanks to the "notify_on_event", whenever there is a new registration 
(see E_UL_CONTACT_INSERT) for that user (see the $avp(filter)), the 
"fork_call" route will be execute and a new branch (for that call) will 
be create toward the new registration - see the t_inject_branches() docs.


Regards,

Bogdan-Andrei Iancu

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

On 15.01.2024 23:12, r...@rvgeerligs.nl wrote:

Hi openSIPS team,

I do not understand the documentaion (opensips version 3.4 manual) on 
the event_routing module.


The example given 1.6.1 push notification does not seem to do just that.

It says:
"
1.6.1. |Push Notification|

We use/notify_on_event/to capture the events on new contact 
registrations for callee. Once the call is sent to callee, based on 
the notification (for new contacts) we inject the newly registered 
contacts as new branches in the ongoing transaction.


Schematics : when we send a call to a user, we subscribe to see any 
new contacts being registered by the user. On such a notification, we 
add the new contact as a new branch to the ongoing transaction 
(ringing) to user.


"
It seems this has to do with notify_on_event.
It may be that the new branches are be added with device information 
for the push notification, but I do not see that in the example given.


Please advise,

Regards,

Ronald Geerligs

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Re: [OpenSIPS-Users] opensips 3.4 documentation event_routing example 1.6.1 - push notification

[Ihor Olkhovskyi]

Ronald,

Have you checked https://www.youtube.com/watch?v=Bx0_l0ONa0g ?

Le lun. 15 janv. 2024 à 22:14,  a écrit :

> Hi openSIPS team,
>
> I do not understand the documentaion (opensips version 3.4 manual) on the
> event_routing module.
>
> The example given 1.6.1 push notification does not seem to do just that.
>
> It says:
> "
> 1.6.1. Push Notification
>
> We use*notify_on_event*to capture the events on new contact registrations
> for callee. Once the call is sent to callee, based on the notification (for
> new contacts) we inject the newly registered contacts as new branches in
> the ongoing transaction.
>
> Schematics : when we send a call to a user, we subscribe to see any new
> contacts being registered by the user. On such a notification, we add the
> new contact as a new branch to the ongoing transaction (ringing) to user.
> "
> It seems this has to do with notify_on_event.
> It may be that the new branches are be added with device information for
> the push notification, but I do not see that in the example given.
>
> Please advise,
>
> Regards,
>
> Ronald Geerligs
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>


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Ihor (Igor)
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[OpenSIPS-Users] opensips 3.4 documentation event_routing example 1.6.1 - push notification

[rvg]

Hi openSIPS team,

I do not understand the documentaion (opensips version 3.4 manual) on the 
event_routing module.

The example given 1.6.1 push notification does not seem to do just that.

It says:
"
1.6.1. Push Notification


We use*notify_on_event*to capture the events on new contact registrations for 
callee. Once the call is sent to callee, based on the notification (for new 
contacts) we inject the newly registered contacts as new branches in the 
ongoing transaction.



Schematics : when we send a call to a user, we subscribe to see any new 
contacts being registered by the user. On such a notification, we add the new 
contact as a new branch to the ongoing transaction (ringing) to user.

"
It seems this has to do with notify_on_event.
It may be that the new branches are be added with device information for the 
push notification, but I do not see that in the example given.

Please advise,

Regards,

Ronald Geerligs___
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[OpenSIPS-Users] [OpenSIPS-Devel] [Release] OpenSIPS 3.4.3, 3.3.9 and 3.2.16 Minor Releases; EoL for OpenSIPS 3.3!

[Liviu Chircu]

Hello!

It is my pleasure to announce a new round of stable minor releases: 
*3.4.3*, *3.3.9* and *3.2.16*.


Do make sure to schedule an update, as they contain important fixes, to 
name a few:
  * dialog: fix PKG memory fragmentation issues around 'dlg_list' (a 
long-awaited fix for a mysterious, constantly-recurring problem)
  * rest_client: async timeouts now work as intended;  dangling 
requests, due to no response from server, should no longer be possible 
and should always time out
  * B2B: fixes on UAC side (fix To URI vs. R-URI construction;  fix BYE 
generation for dangling entities)

  * drouting: allow rule fallback even across prefixless rules (prefix: "")

Note that OpenSIPS *3.3* has reached its *end-of-life*, according to the 
release policy .  It 
will no longer receive any more fixes, with *3.2 LTS *and *3.4 LTS* 
being the currently supported stable versions.


Full changelogs:

https://opensips.org/pub/opensips/3.4.3/ChangeLog
https://opensips.org/pub/opensips/3.3.9/ChangeLog
https://opensips.org/pub/opensips/3.2.16/ChangeLog

Happy Holidays!
OpenSIPS Team

On 27.11.2023 18:07, Liviu Chircu wrote:


Hi, everyone!

The 3.4.3, 3.3.9 and 3.2.16 OpenSIPS minor versions are scheduled for 
release on *Wednesday, Dec 20th*.


In preparation for the releases, starting *Wednesday, Dec 6th*, we 
will impose the usual *freeze* on any significant fixes (as 
complexity) on these stable branches, in order to ensure a /two-week 
safe window/ for testing.


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


Best regards,

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

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[OpenSIPS-Users] OpenSIPS Summit 2024 - Valencia, Spain

[Bogdan-Andrei Iancu]

 OpenSIPS Summit


 May 14th - 17th, 2024

Valencia, Spain


*Bridging people, bridging technologies, bridging experiences
*

The OpenSIPS Summit 
 is the meeting 
place for the OpenSIPS community, for experts, developers and users from 
all over the world, looking to learn and gain knowledge. The OpenSIPS 
Summit is a melting pot for discussion on new technology, for sharing 
experiences, for brainstorming on new trends, for building bridges in 
the Open-Source VoIP & RTC ecosystem.


*Some Great Reasons to Attend*

 * Access the latest news, knowledge and experience in the VoIP & RTC world
 * Learn about upcoming 3.5 OpenSIPS release and how you can leverage it
 * Attend unique presentations and interactive technical workshops
 * Meet FOSS developers and community to share experience and comments
 * Get solutions consultancy during the Free Design Clinics
 * Become an Expert attending the OpenSIPS Advanced Training


*Summit Agenda*

 * Two full days of presentations given by key speakers
 * Open Discussions with key people from OpenSIPS and other OSS projects
 * One full day of Interactive Demos and Showcases
 * One full day of Design Clinics to validate your OpenSIPS deployments
 * One full day OpenSIPS Training (limited seats!)
 * Social events in the beautiful Valencia


*Attend to learn* - the registration process is already open, for both 
online and in-person participants. Note that the training and Design 
Clinics options are available only for the in-person participants. 
The/*Corporate Package*/ is available with an attractive discount.


Register now 

*Speak to share* - the Call for Papers is open for in-person and online 
speakers. Our speaker will enjoy free admission to the event, covering 
lunches and evening events.


Submit your paper now 

*Sponsor to help* - we welcome any help in making the Summit such a 
great event. Sponsoring is a natural way of saying "Thank you" for the 
Open Source code you are using within your businesses.


Become a sponsor 



Interested? Please contact 
 our 
team or email  us!



**









--
Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] opensips and NAT

[rvg]

Hi,

Should I try something like

 nat_uac_test(diff-ip-src-contact || private-contact || diff-ip-src-via || 
diff-port-src-via)?

Any ideas?

Regards,

Ronald




December 15, 2023 at 4:38 PM, r...@rvgeerligs.nl wrote:


> 
> Hi
> 
> I use opensips 3.4 and I have NAT problems with 2 devices behind the same NAT 
> (called party hears nothing).
> The A party is softphone on iPhone (linphone) the B (called)party is 
> Polycom310. The other way around works (Polycom calls linphone).
> 
> Actually I tested this in two different NAT locations. 
> The first location repeatedly works. The second location gives the problem. 
> Both locations have changing public IP addresses and DHCP on 192.168 network.
> Both locations have one FX number assigned to them. No SIP ALG active. 
> 
> I use nat_uac_test(diff-ip-src-contact).
> 
> There is a table:
> 1.5.5.  nat_uac_test(flags)
> Tries to guess if client's request originated behind a nat. The parameter 
> determines what heuristics is used.
> 
> Meaning of the flags (string) parameter is as follows:
> 
> private-contact - (old 1 flag) Contact header field is searched for 
> occurrence of RFC1918 / RFC6598 addresses.
> 
> diff-ip-src-via - (old 2 flag) the "received" test is used: address in Via is 
> compared against source IP address of signaling
> 
> private-via - (old 4 flag) Top Most VIA is searched for occurrence of RFC1918 
> / RFC6598 addresses
> 
> private-sdp - (old 8 flag) SDP is searched for occurrence of RFC1918 / 
> RFC6598 addresses
> 
> diff-port-src-via - (old 16 flag) test if the source port is different from 
> the port in Via
> 
> diff-ip-src-contact - (old 32 flag) address in Contact is compared against 
> source IP address of signaling. 
> 
> diff-port-src-contact - (old 64 flag) Port in Contact is compared against 
> source port of signaling 
> 
> carrier-grade-nat - (old 128 flag) also include RFC 6333 addresses in the 
> checks for ct, via and sdp flags.
> 
> A CSV of the above flags can be provided, the test returns true if any of the 
> tests identified a NAT.
> 
> Currently I use old flag 32.
> 
> I read that using the equivalent of 19 might help but I dont see that in the 
> table.
> 
> Any advice is appreciated.
> 
> Regards,
> 
> Ronald Geerligs
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[OpenSIPS-Users] opensips and NAT

[rvg]

Hi

I use opensips 3.4 and I have NAT problems with 2 devices behind the same NAT 
(called party hears nothing).
The A party is softphone on iPhone (linphone) the B (called)party is 
Polycom310. The other way around works (Polycom calls linphone).

Actually I tested this in two different NAT locations. 
The first location repeatedly works. The second location gives the problem. 
Both locations have changing public IP addresses and DHCP on 192.168 network.
Both locations have one FX number assigned to them. No SIP ALG active. 

I use nat_uac_test(diff-ip-src-contact).

There is a table:
1.5.5.  nat_uac_test(flags)
Tries to guess if client's request originated behind a nat. The parameter 
determines what heuristics is used.

Meaning of the flags (string) parameter is as follows:

private-contact - (old 1 flag) Contact header field is searched for occurrence 
of RFC1918 / RFC6598 addresses.

diff-ip-src-via - (old 2 flag) the "received" test is used: address in Via is 
compared against source IP address of signaling

private-via - (old 4 flag) Top Most VIA is searched for occurrence of RFC1918 / 
RFC6598 addresses

private-sdp - (old 8 flag) SDP is searched for occurrence of RFC1918 / RFC6598 
addresses

diff-port-src-via - (old 16 flag) test if the source port is different from the 
port in Via

diff-ip-src-contact - (old 32 flag) address in Contact is compared against 
source IP address of signaling. 

diff-port-src-contact - (old 64 flag) Port in Contact is compared against 
source port of signaling 

carrier-grade-nat - (old 128 flag) also include RFC 6333 addresses in the 
checks for ct, via and sdp flags.

A CSV of the above flags can be provided, the test returns true if any of the 
tests identified a NAT.


Currently I use old flag 32.

I read that using the equivalent of 19 might help but I dont see that in the 
table.


Any advice is appreciated.

Regards,


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Re: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question

[Bogdan-Andrei Iancu]

Hi Ronald,

Thanks for troubleshooting and fixing this. I pushed the fix on git !
https://github.com/OpenSIPS/opensips-cp/commit/492a0eb46dedc1f09297ed331013254cdd5bf1a8

Best regards,

Bogdan-Andrei Iancu

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

On 08.12.2023 16:57, r...@rvgeerligs.nl wrote:

Hi Bogan-Andrei,

Show errors was not an otion: I get error after log in. Instead I did 
the following:


using tail -f on apache error log file


calling call center on the left menu and result is blank page

[..]


calling TLS management on the left panel and result is blank page

[..]


Calling UAC Registrant on the left menu and result is blank page

[..]


--

diff /var/www/html/opensipscp/web/common/tools/tviewer/tviewer.php 
/var/www/html/opensips-cp/web/common/tools/tviewer/tviewer.php



68,72c68,70
< if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns']) 
{

< for ($i=0; $i
< if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']) 
&& 
$custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']!="" 
&& 
file_exists($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']))
< 
require($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']);

< }
---
> for ($i=0; 
$i$i++) {
> if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']) 
&& 
$custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']!="" 
&& 
file_exists($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']))
> 
require($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_columns'][$i]['action_script']);

85,89c83,85
< if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'])) 
{

< for ($i=0; $i
< if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']) 
&& 
$custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']!="" 
&& 
file_exists($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']))
< 
require($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']);

< }
---
> for ($i=0; 
$i$i++) {
> if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']) 
&& 
$custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']!="" 
&& 
file_exists($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']))
> 
require($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_action_buttons'][$i]['action_script']);

98,99c94
< if 
(isset($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_search']['enabled']) 
&&
< 
$custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_search']['enabled'])

---
> if 
($custom_config[$module_id][$_SESSION[$module_id]['submenu_item_id']]['custom_search']['enabled'])



I hope this is enough?

Regards,

Ronald

December 8, 2023 at 8:39 AM, "Bogdan-Andrei Iancu" 
> 
wrote:


Hi Ronald,

yes, please check for ERR or WARN in php.

Regards,

Bogdan-Andrei Iancu

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

On 07.12.2023 22:26, r...@rvgeerligs.nl wrote:

Hi Bogdan-Andrei,
The others like adresses and dialplan or dynamic routing (very
important one) work and the dialogues give the right answers.
No reaction means blank page.
Want me to enable PHP error messages?
Regards,
Ronald

December 7, 2023 at 6:18 PM, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org?to=%22Bogdan-Andrei%20Iancu%22%20%3Cbogdan%40opensips.org%3E>>
wrote:

Hi Ronald,

Tool as Addresses or Diaplan are still working in 9.3.4 ?
and just to clarify, what exactly means "no reaction"?
when you click on the "Call Center" tool, you still have
the old tool in the main right window? or is it turning
blank ?

Regards,

Bogdan-Andrei Iancu

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

Re: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question

[Bogdan-Andrei Iancu]

Hi Ronald,

yes, please check for ERR or WARN in php.

Regards,

Bogdan-Andrei Iancu

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

On 07.12.2023 22:26, r...@rvgeerligs.nl wrote:

Hi Bogdan-Andrei,

The others like adresses and dialplan or dynamic routing (very 
important one) work and the dialogues give the right answers.

No reaction means blank page.

Want me to enable PHP error messages?

Regards,

Ronald

December 7, 2023 at 6:18 PM, "Bogdan-Andrei Iancu" 
> 
wrote:


Hi Ronald,

Tool as Addresses or Diaplan are still working in 9.3.4 ? and just
to clarify, what exactly means "no reaction"? when you click on
the "Call Center" tool, you still have the old tool in the main
right window? or is it turning blank ?

Regards,

Bogdan-Andrei Iancu

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

On 07.12.2023 15:44, r...@rvgeerligs.nl wrote:

Hi Bogdan,
Short answer: yes.
Did not change anything in modules.inc.php.
I have 2 directories opensips-cp and opensipscp, containing
9.3.3 and 9.3.4 respectively. also configured in apache2.
Copied the db.inc.php from the 9.3.3 to the 9.3.4 directory.
Regards,
Ronald Geerligs

December 7, 2023 at 9:08 AM, "Bogdan-Andrei Iancu"
mailto:bog...@opensips.org?to=%22Bogdan-Andrei%20Iancu%22%20%3Cbogdan%40opensips.org%3E>>
wrote:

Hi Ronald,

So same server with an OpenSIPS 3.4 works with OCP 9.3.3,
but not with OCP 9.3.4 - the only difference between the 2
tests is the version of OCP, still exactly the same env is
used, right ? Any setting in the modules.inc.php ?

Regards,

Bogdan-Andrei Iancu

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

On 06.12.2023 20:58, r...@rvgeerligs.nl wrote:

Hi all,
I do not know where to post this, but maybe it is all
right.
I use:
Opensips 3.4 compiled on Oracle VM ARM Ubuntu 22.04.2
LTS - php 7.4 with OCP 9.3.3. Everything works!
Installed OCP 9.3.4 and many menues on the left side
have no reaction:
CallCenter
keepalived
TLS management !
UAC registrant
SMPP Gateway
TCP Management
Regards,
Ronald Geerligs

October 2, 2023 at 8:49 AM, "Răzvan Crainea"
mailto:raz...@opensips.org?to=%22R%C4%83zvan%20Crainea%22%20%3Crazvan%40opensips.org%3E>>
wrote:

Hi, Nineto!
Although it was not already released, OpenSIPS
master branch should be
compatible with OpenSIPS 3.4. The compatibility
process is not yet
complete, therefore a full release (9.3.4) is not
available yet for
OpenSIPS 3.4.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com/ /
https://www.siphub.com/
On 9/29/23 19:15, Nine to one wrote:

Hello OpenSIPS Control Panel developers,
From website OCP only mentioned support up to
OpenSIPS 3.3, I am using
3.4, so want to know if current OCP already
support OpenSIPS 3.4 or not.
Thanks,
Nineto
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Re: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question

[Bogdan-Andrei Iancu]

Hi Ronald,

Tool as Addresses or Diaplan are still working in 9.3.4 ? and just to 
clarify, what exactly means "no reaction"? when you click on the "Call 
Center" tool, you still have the old tool in the main right window? or 
is it turning blank ?


Regards,

Bogdan-Andrei Iancu

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

On 07.12.2023 15:44, r...@rvgeerligs.nl wrote:

Hi Bogdan,

Short answer: yes.
Did not change anything in modules.inc.php.

I have 2 directories opensips-cp and opensipscp, containing 9.3.3 and 
9.3.4 respectively. also configured in apache2.

Copied the db.inc.php from the 9.3.3 to the 9.3.4 directory.

Regards,

Ronald Geerligs

December 7, 2023 at 9:08 AM, "Bogdan-Andrei Iancu" 
> 
wrote:


Hi Ronald,

So same server with an OpenSIPS 3.4 works with OCP 9.3.3, but not
with OCP 9.3.4 - the only difference between the 2 tests is the
version of OCP, still exactly the same env is used, right ? Any
setting in the modules.inc.php ?

Regards,

Bogdan-Andrei Iancu

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

On 06.12.2023 20:58, r...@rvgeerligs.nl wrote:

Hi all,
I do not know where to post this, but maybe it is all right.
I use:
Opensips 3.4 compiled on Oracle VM ARM Ubuntu 22.04.2 LTS -
php 7.4 with OCP 9.3.3. Everything works!
Installed OCP 9.3.4 and many menues on the left side have no
reaction:
CallCenter
keepalived
TLS management !
UAC registrant
SMPP Gateway
TCP Management
Regards,
Ronald Geerligs

October 2, 2023 at 8:49 AM, "Răzvan Crainea"
mailto:raz...@opensips.org?to=%22R%C4%83zvan%20Crainea%22%20%3Crazvan%40opensips.org%3E>>
wrote:

Hi, Nineto!
Although it was not already released, OpenSIPS master
branch should be
compatible with OpenSIPS 3.4. The compatibility process is
not yet
complete, therefore a full release (9.3.4) is not
available yet for
OpenSIPS 3.4.
Best regards,
Răzvan Crainea
OpenSIPS Core Developer / SIPhub CTO
http://www.opensips-solutions.com/ / https://www.siphub.com/
On 9/29/23 19:15, Nine to one wrote:

Hello OpenSIPS Control Panel developers,
From website OCP only mentioned support up to OpenSIPS
3.3, I am using
3.4, so want to know if current OCP already support
OpenSIPS 3.4 or not.
Thanks,
Nineto
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Re: [OpenSIPS-Users] opensips is restarting after call is finished

[Bogdan-Andrei Iancu]

Hi Simon,

Well, the log is very self explanatory:
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
child process 28495 exited by a signal 11
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
core was generated
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
terminating due to SIGCHLD


Or shortly, you opensips just crashed :). So see 
https://opensips.org/Documentation/TroubleShooting-Crash for how to 
report further.


Regards,

Bogdan-Andrei Iancu

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

On 07.12.2023 17:17, Simon Gajski via Users wrote:


Hi

we are running opensips 3.4.2
on Ubuntu 22.04.3 LTS
and also use RTPengine Version: 10.5.0.0+0~mr10.5.0.0 git-master-74075f63

Opensips acts as SBC with RTP engine enabled.

Calls are working fine, however after each call is finished, bellow 
action happens.


We had same problem with opensips 3.2 and Ubuntu 20. So we did upgrade 
to latest stable release, and it is the same.


Is this opensips script configuration issue or a bug? And how could I 
fix it?


Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
child process 28495 exited by a signal 11
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
core was generated
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
terminating due to SIGCHLD
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28491]: INFO:core:sig_usr: 
signal 15 received
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28492]: INFO:core:sig_usr: 
signal 15 received
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 4(28492) [timer] terminated, 
still waiting for 16 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 18(28508) [TCP main] terminated, 
still waiting for 15 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 3(28491) [time_keeper] 
terminated, still waiting for 14 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 16(28504) [TCP receiver] 
terminated, still waiting for 13 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 2(28490) [MI FIFO] terminated, 
still waiting for 12 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 15(28503) [TCP receiver] 
terminated, still waiting for 11 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 14(28502) [TCP receiver] 
terminated, still waiting for 10 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 12(28500) [TCP receiver] 
terminated, still waiting for 9 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 1(28489) [HTTPD 127.0.0.1:] 
terminated, still waiting for 8 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 8(28496) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 7 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 10(28498) [TCP receiver] 
terminated, still waiting for 6 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 13(28501) [TCP receiver] 
terminated, still waiting for 5 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 9(28497) [TCP receiver] 
terminated, still waiting for 4 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 11(28499) [TCP receiver] 
terminated, still waiting for 3 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 6(28494) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 2 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 17(28507) [Timer handler] 
terminated, still waiting for 1 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 5(28493) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 0 more

Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:cleanup: cleanup
Dec  7 15:59:47 sbc2 opensips: INFO:core:fix_poll_method: using epoll 
as the IO watch method (auto detected)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: 
version: opensips 3.4.2 (x86_64/linux)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: 
using 64 MB of shared memory, allocator: F_MALLOC
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: 
using 4 MB of private process memory, allocator: F_MALLOC
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
WARNING:core:init_reactor_size: shrinking reactor size from 262144 
(autodetected via rlimit) to 10485 (limited by memory of 10% from 4Mb)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 

[OpenSIPS-Users] opensips is restarting after call is finished

[Simon Gajski via Users]

Hi

we are running opensips 3.4.2
on Ubuntu 22.04.3 LTS
and also use RTPengine Version: 10.5.0.0+0~mr10.5.0.0 git-master-74075f63

Opensips acts as SBC with RTP engine enabled.

Calls are working fine, however after each call is finished, bellow 
action happens.


We had same problem with opensips 3.2 and Ubuntu 20. So we did upgrade 
to latest stable release, and it is the same.


Is this opensips script configuration issue or a bug? And how could I 
fix it?


Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
child process 28495 exited by a signal 11
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
core was generated
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:handle_sigs: 
terminating due to SIGCHLD
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28491]: INFO:core:sig_usr: 
signal 15 received
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28492]: INFO:core:sig_usr: 
signal 15 received
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 4(28492) [timer] terminated, still 
waiting for 16 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 18(28508) [TCP main] terminated, 
still waiting for 15 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 3(28491) [time_keeper] terminated, 
still waiting for 14 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 16(28504) [TCP receiver] 
terminated, still waiting for 13 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 2(28490) [MI FIFO] terminated, 
still waiting for 12 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 15(28503) [TCP receiver] 
terminated, still waiting for 11 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 14(28502) [TCP receiver] 
terminated, still waiting for 10 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 12(28500) [TCP receiver] 
terminated, still waiting for 9 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 1(28489) [HTTPD 127.0.0.1:] 
terminated, still waiting for 8 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 8(28496) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 7 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 10(28498) [TCP receiver] 
terminated, still waiting for 6 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 13(28501) [TCP receiver] 
terminated, still waiting for 5 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 9(28497) [TCP receiver] terminated, 
still waiting for 4 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 11(28499) [TCP receiver] 
terminated, still waiting for 3 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 6(28494) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 2 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 17(28507) [Timer handler] 
terminated, still waiting for 1 more
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: 
INFO:core:shutdown_opensips: process 5(28493) [SIP receiver 
udp:213.253.120.65:5060] terminated, still waiting for 0 more

Dec  7 15:59:47 sbc2 /usr/sbin/opensips[28488]: INFO:core:cleanup: cleanup
Dec  7 15:59:47 sbc2 opensips: INFO:core:fix_poll_method: using epoll as 
the IO watch method (auto detected)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: 
version: opensips 3.4.2 (x86_64/linux)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: using 
64 MB of shared memory, allocator: F_MALLOC
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: NOTICE:core:main: using 
4 MB of private process memory, allocator: F_MALLOC
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
WARNING:core:init_reactor_size: shrinking reactor size from 262144 
(autodetected via rlimit) to 10485 (limited by memory of 10% from 4Mb)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
WARNING:core:init_reactor_size: use 'open_files_limit' to enforce other 
limit or increase pkg memory
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
INFO:core:init_reactor_size: reactor size 10485 (using up to 0.40Mb of 
memory per process)
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
INFO:core:evi_publish_event: Registered event 
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
INFO:core:evi_publish_event: Registered event 
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
INFO:core:evi_publish_event: Registered event 
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 
INFO:core:evi_publish_event: Registered event 
Dec  7 15:59:47 sbc2 /usr/sbin/opensips[31289]: 

Re: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question

[Bogdan-Andrei Iancu]

Hi Ronald,

So same server with an OpenSIPS 3.4 works with OCP 9.3.3, but not with 
OCP 9.3.4 - the only difference between the 2 tests is the version of 
OCP, still exactly the same env is used, right ? Any setting in the 
modules.inc.php ?


Regards,

Bogdan-Andrei Iancu

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

On 06.12.2023 20:58, r...@rvgeerligs.nl wrote:

Hi all,

I do not know where to post this, but maybe it is all right.

I use:
Opensips 3.4 compiled on Oracle VM ARM Ubuntu 22.04.2 LTS - php 7.4 
with OCP 9.3.3. Everything works!

Installed OCP 9.3.4 and many menues on the left side have no reaction:
CallCenter
keepalived
TLS management !
UAC registrant
SMPP Gateway
TCP Management

Regards,

Ronald Geerligs


October 2, 2023 at 8:49 AM, "Răzvan Crainea" > 
wrote:


Hi, Nineto!

Although it was not already released, OpenSIPS master branch
should be
compatible with OpenSIPS 3.4. The compatibility process is not yet
complete, therefore a full release (9.3.4) is not available yet for
OpenSIPS 3.4.

Best regards,

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

On 9/29/23 19:15, Nine to one wrote:

Hello OpenSIPS Control Panel developers,
From website OCP only mentioned support up to OpenSIPS 3.3, I
am using
3.4, so want to know if current OCP already support OpenSIPS
3.4 or not.
Thanks,
Nineto
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Re: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question

[rvg]

Hi all,

I do not know where to post this, but maybe it is all right.

I use:
Opensips 3.4 compiled on Oracle VM ARM Ubuntu 22.04.2 LTS - php 7.4 with OCP 
9.3.3. Everything works!
Installed OCP 9.3.4 and many menues on the left side have no reaction:
CallCenter
keepalived
TLS management !
UAC registrant
SMPP Gateway
TCP Management

Regards,

Ronald Geerligs




October 2, 2023 at 8:49 AM, "Răzvan Crainea"  wrote:


> 
> Hi, Nineto!
> 
> Although it was not already released, OpenSIPS master branch should be 
> compatible with OpenSIPS 3.4. The compatibility process is not yet 
> complete, therefore a full release (9.3.4) is not available yet for 
> OpenSIPS 3.4.
> 
> Best regards,
> 
> Răzvan Crainea
> OpenSIPS Core Developer / SIPhub CTO
> http://www.opensips-solutions.com/ / https://www.siphub.com/
> 
> On 9/29/23 19:15, Nine to one wrote:
> 
> > 
> > Hello OpenSIPS Control Panel developers,
> >  
> >  From website OCP only mentioned support up to OpenSIPS 3.3, I am using 
> >  3.4, so want to know if current OCP already support OpenSIPS 3.4 or not.
> >  
> >  Thanks,
> >  Nineto
> >  
> >  ___
> >  Users mailing list
> >  Users@lists.opensips.org
> >  http://lists.opensips.org/cgi-bin/mailman/listinfo/users
> > 
> 
> ___
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Re: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location

[Bogdan-Andrei Iancu]

The people has spoken, *Valencia* will be the city to host OpenSIPS 
Summit 2024:





This is possible thanks to the great help provided by the QXIP team, as 
our local organizer. Now we have to work out the details, we will update 
ASAP.


Best regards,

Bogdan-Andrei Iancu

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

On 11/16/23 2:55 PM, Bogdan-Andrei Iancu wrote:

Dear OpenSIPS'ers,

Thanks to the support and effort of the community, we have some really 
good nominee cities for OpenSIPS Summit 2024.


So here is the list of validated cities:
    https://bit.ly/summit-2024-location-poll

Please help us with the final decision of the event location. Your 
opinion matters to us, so please help us to build a great event for you.


Best regards,



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Re: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location

[Social Boh]

I can't vote without a account or register... maybe would be better vote 
without login.


Regards

---
I'm SoCIaL, MayBe

El 16/11/2023 a las 7:55 a. m., Bogdan-Andrei Iancu escribió:

Dear OpenSIPS'ers,

Thanks to the support and effort of the community, we have some really 
good nominee cities for OpenSIPS Summit 2024.


So here is the list of validated cities:
    https://bit.ly/summit-2024-location-poll

Please help us with the final decision of the event location. Your 
opinion matters to us, so please help us to build a great event for you.


Best regards,



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Re: [OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location

[Bogdan-Andrei Iancu]

In terms of easiness, I agree, but in terms of fairness, I do not :) - 
we need a way to guarantee one vote per person, otherwise the whole 
process will be irrelevant.


Best regards,

Bogdan-Andrei Iancu

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

On 11/16/23 3:13 PM, Social Boh wrote:
I can't vote without a account or register... maybe would be better 
vote without login.


Regards

---
I'm SoCIaL, MayBe

El 16/11/2023 a las 7:55 a. m., Bogdan-Andrei Iancu escribió:

Dear OpenSIPS'ers,

Thanks to the support and effort of the community, we have some 
really good nominee cities for OpenSIPS Summit 2024.


So here is the list of validated cities:
    https://bit.ly/summit-2024-location-poll

Please help us with the final decision of the event location. Your 
opinion matters to us, so please help us to build a great event for you.


Best regards,




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[OpenSIPS-Users] OpenSIPS Summit 2024, Vote for Location

[Bogdan-Andrei Iancu]

Dear OpenSIPS'ers,

Thanks to the support and effort of the community, we have some really 
good nominee cities for OpenSIPS Summit 2024.


So here is the list of validated cities:
    https://bit.ly/summit-2024-location-poll

Please help us with the final decision of the event location. Your 
opinion matters to us, so please help us to build a great event for you.


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] [OpenSIPS-Business] Introducing OpenSIPS 3.5

[Bogdan-Andrei Iancu]

Honored to have your support Giovanni :). We will allocate couple of 
days to (1) start a wiki page with the basic /starting set of 
requirements and (b) let people subscribe so we can a good pool of brains :)


Best regards,

Bogdan-Andrei Iancu

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

On 11/14/23 4:44 PM, Giovanni Maruzzelli wrote:
On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu 
mailto:bog...@opensips.org>> wrote:



This year we introduce a new concept of an *OpenSIPS Working
Group*. And the *IMS OpenSIPS Working Group*
 is
the first one, aiming to gather people with inters in IMS with the
goal to draft, design and implement the IMS support in OpenSIPS.

More details on this may be found here, please read and act:

https://www.opensips.org/Development/Opensips-3-5-Planning





G R E A T !!
(count me on)






--
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18



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Re: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5

[Johan De Clercq]

Me too.


On Tue, 14 Nov 2023, 19:02 Giovanni Maruzzelli,  wrote:

> On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu 
> wrote:
>
>>
>> This year we introduce a new concept of an *OpenSIPS Working Group*. And
>> the *IMS OpenSIPS Working Group*
>>  is the
>> first one, aiming to gather people with inters in IMS with the goal to
>> draft, design and implement the IMS support in OpenSIPS.
>>
>> More details on this may be found here, please read and act:
>>
>> https://www.opensips.org/Development/Opensips-3-5-Planning
>>
>>
>
>
> G R E A T !!
> (count me on)
>
>
>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> OpenTelecom.IT
> cell: +39 347 266 56 18
>
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Re: [OpenSIPS-Users] [OpenSIPS-Business] Introducing OpenSIPS 3.5

[Giovanni Maruzzelli]

On Tue, Nov 14, 2023 at 3:34 PM Bogdan-Andrei Iancu 
wrote:

>
> This year we introduce a new concept of an *OpenSIPS Working Group*. And
> the *IMS OpenSIPS Working Group*
>  is the
> first one, aiming to gather people with inters in IMS with the goal to
> draft, design and implement the IMS support in OpenSIPS.
>
> More details on this may be found here, please read and act:
>
> https://www.opensips.org/Development/Opensips-3-5-Planning
>
>


G R E A T !!
(count me on)






-- 
Sincerely,

Giovanni Maruzzelli
OpenTelecom.IT
cell: +39 347 266 56 18
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Bogdan-Andrei Iancu]

Hi there,

trying to maintain a dialog stateful UAS from script level may be 
something difficult and painful to do. Maybe you should take a look at 
the UAC/UAS support provided by the b2b_entities module in OpenSIPS 3.4:

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 11/11/23 3:26 AM, Kevin Kennedy wrote:
I was able to send the BYE to the call by adding a parameter in the 
dialog module to timeout the dialog with a short time letting the 
announcement play, and added the create_dialog with the flag of B to 
send BYE on dialog timeout at the beginning of the route.  Now that 
the transactions are working correctly, I can use the same route for 
the calls with SDP as well and tighten up the script.  Thanks for 
helping out with some code examples, and letting me update on my 
progress on this thread.  Hopefully this can help someone else out 
having a similar problem when trying to use Opensips with RTPENGINE as 
an announcement server.


modparam("dialog", "default_timeout", 12)

route["RTPENGINE"]{
    if (has_body("application/sdp")) {
        create_dialog("B");
        rtpengine_offer();
        $json(reply) := $rtpquery;
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
address);

        remove_body_part();
append_to_reply("Contact:\r\n");
        append_to_reply("Content-Type: application/sdp\r\n");
        $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
        $var(body) = $(var(body){re.subst,/(audio.)./\1$var(port)/g});
        t_reply_with_body(200, "OK", $var(body));
        rtpengine_play_media("call-id=$ci from-tag=$ft 
file=/etc/rtpengine/unk_num.wav");

        exit;
    } else {
        create_dialog("B");
        $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 " + 
$socket_in(ip) + "\r\ns=-\r\nc=IN IP4 " + $socket_in(ip) + "\r\nt=0 
0\r\nm=audio " + $sp + " RTP/AVP 0 101\r\na=sendrecv\r\na=rtpmap:0 
PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\n");
        rtpengine_offer("from-tag=$ft replace-session-connection 
trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody));

        $json(reply) := $rtpquery;
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
address);

append_to_reply("Contact:        rtpengine_play_media("call-id=$ci from-tag=$ft 
file=/etc/rtpengine/unk_num.wav");

        exit;
    }

}

Thank you.

Kevin

On Fri, Nov 10, 2023 at 4:54 PM Kevin Kennedy > wrote:


Looks like if I put t_newtran(); in the main route this created
the transaction and allowed the ACK to be recognized.  Now How do
I force Opensips to send a BYE.

Thank you.

On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy
mailto:kennedy4...@gmail.com>> wrote:




I was able to get audio, The
problem I was having is the
Originator string in the SDP. 
However, I am still having the
same issue with accepting the ACK
from the Originator and not
resending the 200OK.  Can someone
please help with this issue?

Thank you

*Code snippet for the Late Media
route*
route["LateMedia3"]{
    if (has_body("application/sdp")) {
        xlog(" Entered
route LateMedia3 with Fake SDP
from Originator \r\n");
rtpengine_offer();
        $json(reply) := $rtpquery;

$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
port);

$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
address);
remove_body_part();

append_to_reply("Contact:\r\n");
append_to_reply("Content-Type:
application/sdp\r\n");
$var(body) =
$(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
        $var(body) =

Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

I was able to send the BYE to the call by adding a parameter in the dialog
module to timeout the dialog with a short time letting the announcement
play, and added the create_dialog with the flag of B to send BYE on dialog
timeout at the beginning of the route.  Now that the transactions are
working correctly, I can use the same route for the calls with SDP as well
and tighten up the script.  Thanks for helping out with some code examples,
and letting me update on my progress on this thread.  Hopefully this can
help someone else out having a similar problem when trying to use Opensips
with RTPENGINE as an announcement server.

modparam("dialog", "default_timeout", 12)

route["RTPENGINE"]{
if (has_body("application/sdp")) {
create_dialog("B");
rtpengine_offer();
$json(reply) := $rtpquery;
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
port);
$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
address);
remove_body_part();
append_to_reply("Contact:\r\n");
append_to_reply("Content-Type: application/sdp\r\n");
$var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
$var(body) = $(var(body){re.subst,/(audio.)./\1$var(port)/g});
t_reply_with_body(200, "OK", $var(body));
rtpengine_play_media("call-id=$ci from-tag=$ft
file=/etc/rtpengine/unk_num.wav");
exit;
} else {
create_dialog("B");
$var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4 " +
$socket_in(ip) + "\r\ns=-\r\nc=IN IP4 " + $socket_in(ip) + "\r\nt=0
0\r\nm=audio " + $sp + " RTP/AVP 0 101\r\na=sendrecv\r\na=rtpmap:0
PCMU/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-15\r\n");
rtpengine_offer("from-tag=$ft replace-session-connection
trust-address replace-origin codec-strip-g729",,$var(body),$var(newbody));
$json(reply) := $rtpquery;
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
port);
$var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
address);
append_to_reply("Contact: wrote:

> Looks like if I put t_newtran(); in the main route this created the
> transaction and allowed the ACK to be recognized.  Now How do I force
> Opensips to send a BYE.
>
> Thank you.
>
> On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy 
> wrote:
>
>>
>>
>>
> I was able to get audio,  The problem I was having is the
> Originator string in the SDP.  However, I am still having the same 
> issue
> with accepting the ACK from the Originator and not resending the 
> 200OK.
> Can someone please help with this issue?
>
> Thank you
>
> *Code snippet for the Late Media route*
> route["LateMedia3"]{
> if (has_body("application/sdp")) {
> xlog(" Entered route LateMedia3 with Fake SDP from
> Originator \r\n");
> rtpengine_offer();
> $json(reply) := $rtpquery;
>
> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
> port);
>
> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
> address);
> remove_body_part();
> append_to_reply("Contact: $socket_in(ip):$socket_in(port);user=phone>\r\n");
> append_to_reply("Content-Type: application/sdp\r\n");
> $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
> $var(body) =
> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
> t_reply_with_body(200, "OK", $var(body));
> rtpengine_play_media("call-id=$ci from-tag=$ft
> file=/etc/rtpengine/unk_num.wav");
> async(sleep(10), after_media);
>  } else {
> xlog(" Entered route LateMedia3 No SDP received,
> Create one from variable \r\n");
> $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4
> 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 
> 3140
> RTP/AVP 0 101\r\na
> =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101
> telephone-event/8000\r\na=fmtp:101 0-15\r\n");
> xlog("# Body to RTPENGINE is
> ###\r\n$var(newbody)\r\n");
> rtpengine_offer("from-tag=$ft replace-session-connection
> trust-address replace-origin 
> codec-strip-g729",,$var(body),$var(newbody));
> xlog("# Body from RTPENGINE is
> ###\r\n$var(body)\r\n");
> $json(reply) := $rtpquery;
>
> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
> port);
>
> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 

Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

Looks like if I put t_newtran(); in the main route this created the
transaction and allowed the ACK to be recognized.  Now How do I force
Opensips to send a BYE.

Thank you.

On Fri, Nov 10, 2023 at 11:44 AM Kevin Kennedy 
wrote:

>
>
>
 I was able to get audio,  The problem I was having is the
 Originator string in the SDP.  However, I am still having the same 
 issue
 with accepting the ACK from the Originator and not resending the 200OK.
 Can someone please help with this issue?

 Thank you

 *Code snippet for the Late Media route*
 route["LateMedia3"]{
 if (has_body("application/sdp")) {
 xlog(" Entered route LateMedia3 with Fake SDP from
 Originator \r\n");
 rtpengine_offer();
 $json(reply) := $rtpquery;

 $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
 port);

 $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
 address);
 remove_body_part();
 append_to_reply("Contact:>>> $socket_in(ip):$socket_in(port);user=phone>\r\n");
 append_to_reply("Content-Type: application/sdp\r\n");
 $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
 $var(body) =
 $(var(body){re.subst,/(audio.)./\1$var(port)/g});
 t_reply_with_body(200, "OK", $var(body));
 rtpengine_play_media("call-id=$ci from-tag=$ft
 file=/etc/rtpengine/unk_num.wav");
 async(sleep(10), after_media);
  } else {
 xlog(" Entered route LateMedia3 No SDP received,
 Create one from variable \r\n");
 $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4
 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 
 3140
 RTP/AVP 0 101\r\na
 =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101
 telephone-event/8000\r\na=fmtp:101 0-15\r\n");
 xlog("# Body to RTPENGINE is
 ###\r\n$var(newbody)\r\n");
 rtpengine_offer("from-tag=$ft replace-session-connection
 trust-address replace-origin 
 codec-strip-g729",,$var(body),$var(newbody));
 xlog("# Body from RTPENGINE is
 ###\r\n$var(body)\r\n");
 $json(reply) := $rtpquery;

 $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
 port);

 $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
 address);
 append_to_reply("Contact:>>> $socket_in(ip):$socket_in(port);transport=udp>\r\n");
 append_to_reply("Content-Type: application/sdp\r\n");
 $var(body) =
 $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g});
 $var(body) =
 $(var(body){re.subst,/(audio.)./\1$var(port)/g});
 xlog("# Body being sent in Reply is
 ##\r\n$var(body)\r\n");
 t_reply_with_body(200, "OK", $var(body));
 rtpengine_play_media("call-id=$ci from-tag=$ft
 file=/etc/rtpengine/unk_num.wav");
 async(sleep(10), after_media);
 }
 }

 route[after_media]
 { if (t_was_cancelled()) {
 rtpengine_delete();
 exit;
 } else {
 rtpengine_delete();
 sl_send_reply(486,"Busy here");
 exit;
 }
 }

>>>
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

>>> I was able to get audio,  The problem I was having is the Originator
>>> string in the SDP.  However, I am still having the same issue with
>>> accepting the ACK from the Originator and not resending the 200OK.  Can
>>> someone please help with this issue?
>>>
>>> Thank you
>>>
>>> *Code snippet for the Late Media route*
>>> route["LateMedia3"]{
>>> if (has_body("application/sdp")) {
>>> xlog(" Entered route LateMedia3 with Fake SDP from
>>> Originator \r\n");
>>> rtpengine_offer();
>>> $json(reply) := $rtpquery;
>>>
>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
>>>
>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
>>> address);
>>> remove_body_part();
>>> append_to_reply("Contact:>> $socket_in(ip):$socket_in(port);user=phone>\r\n");
>>> append_to_reply("Content-Type: application/sdp\r\n");
>>> $var(body) = $(rb{re.subst,/(IP4.).*/\1$var(addr)/g});
>>> $var(body) =
>>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>> t_reply_with_body(200, "OK", $var(body));
>>> rtpengine_play_media("call-id=$ci from-tag=$ft
>>> file=/etc/rtpengine/unk_num.wav");
>>> async(sleep(10), after_media);
>>>  } else {
>>> xlog(" Entered route LateMedia3 No SDP received,
>>> Create one from variable \r\n");
>>> $var(newbody) = ("v=0\r\no=Opensips " + $Ts + " 0 IN IP4
>>> 10.255.100.147\r\ns=-\r\nc=IN IP4 10.255.100.147\r\nt=0 0\r\nm=audio 
>>> 3140
>>> RTP/AVP 0 101\r\na
>>> =sendrecv\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:101
>>> telephone-event/8000\r\na=fmtp:101 0-15\r\n");
>>> xlog("# Body to RTPENGINE is
>>> ###\r\n$var(newbody)\r\n");
>>> rtpengine_offer("from-tag=$ft replace-session-connection
>>> trust-address replace-origin 
>>> codec-strip-g729",,$var(body),$var(newbody));
>>> xlog("# Body from RTPENGINE is
>>> ###\r\n$var(body)\r\n");
>>> $json(reply) := $rtpquery;
>>>
>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
>>>
>>> $var(addr)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local 
>>> address);
>>> append_to_reply("Contact:>> $socket_in(ip):$socket_in(port);transport=udp>\r\n");
>>> append_to_reply("Content-Type: application/sdp\r\n");
>>> $var(body) = $(var(body){re.subst,/(IP4.).*/\1$var(addr)/g});
>>> $var(body) =
>>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>> xlog("# Body being sent in Reply is
>>> ##\r\n$var(body)\r\n");
>>> t_reply_with_body(200, "OK", $var(body));
>>> rtpengine_play_media("call-id=$ci from-tag=$ft
>>> file=/etc/rtpengine/unk_num.wav");
>>> async(sleep(10), after_media);
>>> }
>>> }
>>>
>>> route[after_media]
>>> { if (t_was_cancelled()) {
>>> rtpengine_delete();
>>> exit;
>>> } else {
>>> rtpengine_delete();
>>> sl_send_reply(486,"Busy here");
>>> exit;
>>> }
>>> }
>>>
>>
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

flags=78
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:t_lookup_request: start searching: hash=31608, isACK=0
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:matching_3261: RFC3261 transaction matching failed
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:t_lookup_request: no transaction found
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:run_any_trans_callbacks: trans=0x7f920581abe8, callback type 1, id 0
> entered
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:core:parse_headers: flags=
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:_reply_light: reply sent out. buf=0x7f92094d8218: SIP/2.0 1...,
> shmem=0x7f920581dfa8: SIP/2.0 1
> Nov  7 11:43:56 lab-opensips opensips[4670]: Nov  7 11:43:56 [4670]
> DBG:tm:_reply_light: finished
>
> Thank you.
>
> Kevin
>
> On Mon, Nov 6, 2023 at 2:58 PM Kevin Kennedy 
> wrote:
>
>> I would like to clarify the issue in case its not 100% clear.
>> * Caller sends INVITE with No SDP(Late Media Invite)
>> * Another device in path (B2BUA) receives INVITE and sends dummy SDP to
>> Opensips with just G.711 codec in the Offer
>> * Opensips Creates Dialog and sends 200OK with SDP using
>> t_reply_with_body based on previously provided information.
>> * B2BUA receives 200OK with SDP then sends ACK followed by a Re-INVITE
>> with No SDP back to OpenSips.
>> * Opensips appears to accept the ACK as it doesn't retransmit the 200OK
>> right away as before updated changes.
>> * Opensips sends 100 trying with new CSEQ from Re-INVITE with no SDP
>> * 200OK Loop created
>> * Opensips send 200 OK with old CSEQ
>> * B2BUA sends ACK with old CSEQ
>> * Call times out.
>>
>> No audio sent
>>
>> Thank you
>>
>> Kevin.
>> *
>>
>>
>>
>> On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy 
>> wrote:
>>
>>> I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there
>>> was some re-invite fixes.  Still doesn't seem to resolve this issue.  What
>>> am I missing to handle this correctly?
>>>
>>> Thank you.
>>>
>>> Kevin
>>>
>>> On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy 
>>> wrote:
>>>
>>>> Dmitry,
>>>> Thank you for your response, it does appear to work this way and is
>>>> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly
>>>> with the updated Cseq in the 100 Trying, but the 200 OK (using the
>>>> t_reply_with_body), still has the same Cseq as the initial INVITE.  How can
>>>> I make adjustments for this?
>>>>
>>>> Thank you.
>>>>
>>>> Kevin
>>>>
>>>> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov 
>>>> wrote:
>>>>
>>>>> It turns out that this is no early_media, there were simply successful
>>>>> attempts with 183 Session Progress, which is why there was such a
>>>>> misunderstanding, I’ll attach the snippet code again in plain text:
>>>>> route { if (is_method("INVITE")) { create_dialog(); route(media);
>>>>> exit;
>>>>> } } route[media] { if (has_body("application/sdp")) {
>>>>> rtpengine_offer();
>>>>> } $json(reply) := $rtpquery;
>>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
>>>>> port);
>>>>> remove_body_part(); append_to_reply("Contact:
>>>>> \r\n");
>>>>>
>>>>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>>>>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>>>>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>>>> t_reply_with_body(200, "OK", $var(body));
>>>>> rtpengine_play_media("call-id=$ci from-tag=$ft
>>>>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); }
>>>>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete();
>>>>> exit;
>>>>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }
>>>>>
>>>>> and pined previous posts below :)
>>>>>
>>>>> >
>>>>> --
>>>>> > Message: 2
>>

[OpenSIPS-Users] OpenSIPS Control Panel 9.3.4 released

[Bogdan-Andrei Iancu]

The OpenSIPS Control Panel 9.3.4 is the corresponding version for 
OpenSIPs 3.4.x .


Besides fixes, the main changes in this version are related to the 
compatibility to OpenSIPS 3.4, in `dispatcher` and `dialog` tools.


http://controlpanel.opensips.org/download.php

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] opensips summit

[Bogdan-Andrei Iancu]

Hi Ihor,

If you can provide local assistance in Geneva, maybe you should consider 
submitting this by email...and let's see how it's comparing with the 
other options we get :)


Best regards,

Bogdan-Andrei Iancu

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

On 11/8/23 12:50 PM, Ihor Olkhovskyi wrote:
I can propose Geneva, but I'm afraid people will be chocked with local 
prices.

But can organise a visit to CERN

Le mar. 7 nov. 2023 à 15:19, Johan De Clercq > a écrit :


good plan.  I haven't visited munich in ages.

Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir mailto:s...@h-m.net>>:

Hi,

perhaps Munich would be also a great city for the next
OpenSIPS Summit.
As a native of Munich I could organize recommending hotels or
the social event.

-- 
BRs

Stefan


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--
Best regards,
Ihor (Igor)

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Re: [OpenSIPS-Users] opensips summit

[Ihor Olkhovskyi]

I can propose Geneva, but I'm afraid people will be chocked with local
prices.
But can organise a visit to CERN

Le mar. 7 nov. 2023 à 15:19, Johan De Clercq  a écrit :

> good plan.  I haven't visited munich in ages.
>
> Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir :
>
>> Hi,
>>
>> perhaps Munich would be also a great city for the next OpenSIPS Summit.
>> As a native of Munich I could organize recommending hotels or the social
>> event.
>>
>> --
>> BRs
>> Stefan
>>
>>
>> ___
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
> ___
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>


-- 
Best regards,
Ihor (Igor)
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

happens, it responds correctly
>>> with the updated Cseq in the 100 Trying, but the 200 OK (using the
>>> t_reply_with_body), still has the same Cseq as the initial INVITE.  How can
>>> I make adjustments for this?
>>>
>>> Thank you.
>>>
>>> Kevin
>>>
>>> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov 
>>> wrote:
>>>
>>>> It turns out that this is no early_media, there were simply successful
>>>> attempts with 183 Session Progress, which is why there was such a
>>>> misunderstanding, I’ll attach the snippet code again in plain text:
>>>> route { if (is_method("INVITE")) { create_dialog(); route(media); exit;
>>>> } } route[media] { if (has_body("application/sdp")) {
>>>> rtpengine_offer();
>>>> } $json(reply) := $rtpquery;
>>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
>>>> port);
>>>> remove_body_part(); append_to_reply("Contact:
>>>> \r\n");
>>>>
>>>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>>>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>>>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>>> t_reply_with_body(200, "OK", $var(body));
>>>> rtpengine_play_media("call-id=$ci from-tag=$ft
>>>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); }
>>>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit;
>>>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }
>>>>
>>>> and pined previous posts below :)
>>>>
>>>> > --
>>>> > Message: 2
>>>> > Date: Fri, 3 Nov 2023 16:00:22 +0500
>>>> > From: Dmitry Ponomaryov
>>>> > To:users@lists.opensips.org
>>>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not
>>>> >   absorbing ACK
>>>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8...@gmail.com>
>>>> > Content-Type: text/plain; charset="utf-8"; Format="flowed"
>>>> >
>>>> > Hello everyone, I would like to show my part of the code when playing
>>>> > early media after 200OK, when creating dialogs, I substituted $DLG_did
>>>> > in the contact of my dialog, and received the same $DLG_did for my
>>>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite
>>>> > having already received an ACK response.
>>>> >
>>>> > route {
>>>> >
>>>> > # initial invite
>>>> >
>>>> > if (is_method("INVITE")) {
>>>> >
>>>> > create_dialog();
>>>> >
>>>> > route(early_media);
>>>> >
>>>> > exit;
>>>> >
>>>> > }
>>>> >
>>>> > } route[early_media] { if (has_body("application/sdp")) {
>>>> > rtpengine_manage(); } $json(reply) := $rtpquery;
>>>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
>>>> port);
>>>> > remove_body_part();
>>>> >
>>>> > append_to_reply("Contact:
>>>> > >>> $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n");
>>>> >
>>>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>>>> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>>>> > $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>>> > t_reply_with_body(200, "OK", $var(body));
>>>> > rtpengine_play_media("call-id=$ci from-tag=$ft
>>>> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media);
>>>> }
>>>> > route[after_early_media] { if (t_was_cancelled()) {
>>>> rtpengine_delete();
>>>> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here");
>>>> exit;
>>>> > } }
>>>> >
>>>> > I don’t know if Kevin example was with creating a dialog, but I also
>>>> > noticed this problem through transaction... thanks
>>>> > -- next part --
>>>> > An HTML att

Re: [OpenSIPS-Users] opensips summit

[Johan De Clercq]

good plan.  I haven't visited munich in ages.

Op di 7 nov 2023 om 15:10 schreef Stefan Hofmeir :

> Hi,
>
> perhaps Munich would be also a great city for the next OpenSIPS Summit.
> As a native of Munich I could organize recommending hotels or the social
> event.
>
> --
> BRs
> Stefan
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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Re: [OpenSIPS-Users] opensips summit

[Stefan Hofmeir]

Hi,

perhaps Munich would be also a great city for the next OpenSIPS Summit.
As a native of Munich I could organize recommending hotels or the social event.

-- 
BRs
Stefan


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[OpenSIPS-Users] opensips summit

[Johan De Clercq]

list,
do we have somebody who can arrange something in Vienna ?
looks like a great location to me :-)
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[OpenSIPS-Users] OpenSIPS Summit 2024, Call for Location

[Bogdan-Andrei Iancu]

Hello,

It is time to start planning the 2024 OpenSIPS Summit. And first thing 
to do is to pick up a location.


At this stage we are looking for nominations, for cities (in Europe) and 
local help/support. We need all the help we can get in order to put 
together yet another great edition of this event .


https://blog.opensips.org/2023/11/07/opensips-summit-2024-call-for-location/

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] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

I would like to clarify the issue in case its not 100% clear.
* Caller sends INVITE with No SDP(Late Media Invite)
* Another device in path (B2BUA) receives INVITE and sends dummy SDP to
Opensips with just G.711 codec in the Offer
* Opensips Creates Dialog and sends 200OK with SDP using t_reply_with_body
based on previously provided information.
* B2BUA receives 200OK with SDP then sends ACK followed by a Re-INVITE with
No SDP back to OpenSips.
* Opensips appears to accept the ACK as it doesn't retransmit the 200OK
right away as before updated changes.
* Opensips sends 100 trying with new CSEQ from Re-INVITE with no SDP
* 200OK Loop created
* Opensips send 200 OK with old CSEQ
* B2BUA sends ACK with old CSEQ
* Call times out.

No audio sent

Thank you

Kevin.
*



On Mon, Nov 6, 2023 at 12:54 PM Kevin Kennedy  wrote:

> I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there
> was some re-invite fixes.  Still doesn't seem to resolve this issue.  What
> am I missing to handle this correctly?
>
> Thank you.
>
> Kevin
>
> On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy 
> wrote:
>
>> Dmitry,
>> Thank you for your response, it does appear to work this way and is
>> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly
>> with the updated Cseq in the 100 Trying, but the 200 OK (using the
>> t_reply_with_body), still has the same Cseq as the initial INVITE.  How can
>> I make adjustments for this?
>>
>> Thank you.
>>
>> Kevin
>>
>> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov 
>> wrote:
>>
>>> It turns out that this is no early_media, there were simply successful
>>> attempts with 183 Session Progress, which is why there was such a
>>> misunderstanding, I’ll attach the snippet code again in plain text:
>>> route { if (is_method("INVITE")) { create_dialog(); route(media); exit;
>>> } } route[media] { if (has_body("application/sdp")) { rtpengine_offer();
>>> } $json(reply) := $rtpquery;
>>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
>>> remove_body_part(); append_to_reply("Contact:
>>> \r\n");
>>>
>>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>>> t_reply_with_body(200, "OK", $var(body));
>>> rtpengine_play_media("call-id=$ci from-tag=$ft
>>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); }
>>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit;
>>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }
>>>
>>> and pined previous posts below :)
>>>
>>> > --
>>> > Message: 2
>>> > Date: Fri, 3 Nov 2023 16:00:22 +0500
>>> > From: Dmitry Ponomaryov
>>> > To:users@lists.opensips.org
>>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not
>>> >   absorbing ACK
>>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8...@gmail.com>
>>> > Content-Type: text/plain; charset="utf-8"; Format="flowed"
>>> >
>>> > Hello everyone, I would like to show my part of the code when playing
>>> > early media after 200OK, when creating dialogs, I substituted $DLG_did
>>> > in the contact of my dialog, and received the same $DLG_did for my
>>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite
>>> > having already received an ACK response.
>>> >
>>> > route {
>>> >
>>> > # initial invite
>>> >
>>> > if (is_method("INVITE")) {
>>> >
>>> > create_dialog();
>>> >
>>> > route(early_media);
>>> >
>>> > exit;
>>> >
>>> > }
>>> >
>>> > } route[early_media] { if (has_body("application/sdp")) {
>>> > rtpengine_manage(); } $json(reply) := $rtpquery;
>>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local
>>> port);
>>> > remove_body_part();
>>> >
>>> > append_to_reply("Contact:
>>> > >> $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n");
>>> >
>>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>>> >

Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

I tried updating from Opensips 3.2 to Opensips 3.4.2 as I saw that there
was some re-invite fixes.  Still doesn't seem to resolve this issue.  What
am I missing to handle this correctly?

Thank you.

Kevin

On Fri, Nov 3, 2023 at 12:51 PM Kevin Kennedy  wrote:

> Dmitry,
> Thank you for your response, it does appear to work this way and is
> absorbing the ACK now, but when a Re-INVITE happens, it responds correctly
> with the updated Cseq in the 100 Trying, but the 200 OK (using the
> t_reply_with_body), still has the same Cseq as the initial INVITE.  How can
> I make adjustments for this?
>
> Thank you.
>
> Kevin
>
> On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov 
> wrote:
>
>> It turns out that this is no early_media, there were simply successful
>> attempts with 183 Session Progress, which is why there was such a
>> misunderstanding, I’ll attach the snippet code again in plain text:
>> route { if (is_method("INVITE")) { create_dialog(); route(media); exit;
>> } } route[media] { if (has_body("application/sdp")) { rtpengine_offer();
>> } $json(reply) := $rtpquery;
>> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
>> remove_body_part(); append_to_reply("Contact:
>> \r\n");
>>
>> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>> t_reply_with_body(200, "OK", $var(body));
>> rtpengine_play_media("call-id=$ci from-tag=$ft
>> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); }
>> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit;
>> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }
>>
>> and pined previous posts below :)
>>
>> > --
>> > Message: 2
>> > Date: Fri, 3 Nov 2023 16:00:22 +0500
>> > From: Dmitry Ponomaryov
>> > To:users@lists.opensips.org
>> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not
>> >   absorbing ACK
>> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8...@gmail.com>
>> > Content-Type: text/plain; charset="utf-8"; Format="flowed"
>> >
>> > Hello everyone, I would like to show my part of the code when playing
>> > early media after 200OK, when creating dialogs, I substituted $DLG_did
>> > in the contact of my dialog, and received the same $DLG_did for my
>> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite
>> > having already received an ACK response.
>> >
>> > route {
>> >
>> > # initial invite
>> >
>> > if (is_method("INVITE")) {
>> >
>> > create_dialog();
>> >
>> > route(early_media);
>> >
>> > exit;
>> >
>> > }
>> >
>> > } route[early_media] { if (has_body("application/sdp")) {
>> > rtpengine_manage(); } $json(reply) := $rtpquery;
>> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
>> > remove_body_part();
>> >
>> > append_to_reply("Contact:
>> > > $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n");
>> >
>> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
>> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
>> > $(var(body){re.subst,/(audio.)./\1$var(port)/g});
>> > t_reply_with_body(200, "OK", $var(body));
>> > rtpengine_play_media("call-id=$ci from-tag=$ft
>> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); }
>> > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete();
>> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit;
>> > } }
>> >
>> > I don’t know if Kevin example was with creating a dialog, but I also
>> > noticed this problem through transaction... thanks
>> > -- next part --
>> > An HTML attachment was scrubbed...
>> > URL:<
>> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html
>> >
>> > --
>> >
>> > Message: 1
>> > Date: Thu, 2 Nov 2023 16:32:02 -0700
>> > From: Kevin Kennedy
>> > To: Op

Re: [OpenSIPS-Users] OpenSIPS as websocket client

[Ihor Olkhovskyi]

Seems to be, default timeouts are too low.

By adding

modparam("proto_wss", "wss_handshake_timeout", 500)
modparam("proto_wss", "wss_tls_handshake_timeout", 500)

everything is working.

Le 02/11/2023 à 20:52, Ihor Olkhovskyi a écrit :

Hello,
I'm a bit new (to a recent versions) to OpenSIPS and trying it to act 
as a UDP - WebSocket proxy using it as an outbound proxy in SIP client 
(PJSUA, if it's important)


Currently I'm using 3.4.2 version.
Config is quite simple, not far from default one.
...
socket=udp:0.0.0.0:6051 
socket=wss:0.0.0.0:9443 
...
loadmodule "proto_udp.so"
loadmodule "proto_tls.so"

# WebSocket part
loadmodule "proto_wss.so"

loadmodule "tls_openssl.so"
loadmodule "tls_mgm.so"

modparam("tls_mgm", "client_domain", "localhost")
modparam("tls_mgm", "certificate", 
"[localhost]/etc/ssl/certs/ssl-cert-snakeoil.pem")
modparam("tls_mgm", "private_key", 
"[localhost]/etc/ssl/private/ssl-cert-snakeoil.key")
modparam("tls_mgm", "ca_list", 
"[localhost]/etc/ssl/certs/ca-certificates.crt")

modparam("tls_mgm", "verify_cert", "[localhost]0")
modparam("tls_mgm", "require_cert", "[localhost]0")

...
route[relay] {
    if ($socket_in(proto) == "UDP") {
        $socket_out = "wss:0.0.0.0:9443 ";
    } else {
        $socket_out = "udp:0.0.0.0:6051 ";
    }

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

I'm using most generic self-signed certs and just started to make some 
experiments.
But when I'm trying just forward SIP packets to remote server, I'm 
getting this in the logs


DBG:core:parse_headers: flags=
DBG:proto_wss:proto_wss_send: no open tcp connection found, opening 
new one

DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384
DBG:core:probe_max_sock_buff: using snd buffer of 416 kb
DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 4
DBG:core:print_ip: tcpconn_new: new tcp connection to: 
DBG:core:tcpconn_new: on port 8089, proto 6
DBG:tls_mgm:tls_find_client_domain: found TLS client domain: localhost
DBG:tls_openssl:openssl_tls_conn_init: Creating a whole new ssl connection
DBG:tls_openssl:openssl_tls_conn_init: Setting in CONNECT mode (client)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
ERROR:tls_openssl:openssl_tls_blocking_write: TLS send timeout (100)
ERROR:proto_wss:ws_client_handshake: cannot start handshake
ERROR:proto_wss:ws_connect: cannot complete WebSocket handshake
DBG:core:tcpconn_destroy: destroying connection 0x7f0efb106440, flags 0038
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
NOTICE:tls_openssl:verify_callback: depth = 2, verify success
NOTICE:tls_openssl:verify_callback: depth = 1, verify success
NOTICE:tls_openssl:verify_callback: depth = 0, verify success
INFO:tls_openssl:openssl_tls_connect: New TLS connection to 
:8089 established
DBG:tls_openssl:openssl_tls_connect: new TLS connection to 
:8089 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256
DBG:tls_openssl:openssl_tls_connect: sending socket: 0.0.0.0:37697 

INFO:tls_openssl:tls_dump_cert_info: tls_connect: server TLS 
certificate subject: /CN=*.pbx.company.domain, issuer: 
/C=GB/ST=Greater Manchester/L=Salford/O=Sectigo Limited/CN=Sectigo RSA 
Domain Validation Secure Server CA
INFO:tls_openssl:tls_dump_cert_info: tls_connect: local TLS client 
certificate subject: /CN=localhost, issuer: /CN=localhost

DBG:tls_openssl:openssl_tls_write: write was successful (6 bytes)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
DBG:tls_openssl:openssl_tls_write: write was successful (2 bytes)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
DBG:tls_openssl:openssl_tls_conn_shutdown: first phase of 2-way 
handshake completed succesfuly

ERROR:proto_wss:proto_wss_send: connect failed
ERROR:tm:msg_send: send() to :8089 for proto wss/6 failed
ERROR:tm:t_forward_nonack: sending request failed
DBG:tm:t_relay_to: t_forward_nonack returned error


Server that I'm making connections to is supporting TLS and WSS 
transports. If I'm changing socket type from WSS to TLS, all is 
working, so it's not a TLS certificate issue or something like this.


I'm pretty sure, that I'm missing something obvious, but not really 
getting what.


Would be appreciated for any hints.
--
Best regards,
Ihor (Igor)___
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

Dmitry,
Thank you for your response, it does appear to work this way and is
absorbing the ACK now, but when a Re-INVITE happens, it responds correctly
with the updated Cseq in the 100 Trying, but the 200 OK (using the
t_reply_with_body), still has the same Cseq as the initial INVITE.  How can
I make adjustments for this?

Thank you.

Kevin

On Fri, Nov 3, 2023 at 5:10 AM Dmitry Ponomaryov  wrote:

> It turns out that this is no early_media, there were simply successful
> attempts with 183 Session Progress, which is why there was such a
> misunderstanding, I’ll attach the snippet code again in plain text:
> route { if (is_method("INVITE")) { create_dialog(); route(media); exit;
> } } route[media] { if (has_body("application/sdp")) { rtpengine_offer();
> } $json(reply) := $rtpquery;
> $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
> remove_body_part(); append_to_reply("Contact:
> \r\n");
>
> append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
> $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
> $(var(body){re.subst,/(audio.)./\1$var(port)/g});
> t_reply_with_body(200, "OK", $var(body));
> rtpengine_play_media("call-id=$ci from-tag=$ft
> file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); }
> route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit;
> } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }
>
> and pined previous posts below :)
>
> > --
> > Message: 2
> > Date: Fri, 3 Nov 2023 16:00:22 +0500
> > From: Dmitry Ponomaryov
> > To:users@lists.opensips.org
> > Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not
> >   absorbing ACK
> > Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8...@gmail.com>
> > Content-Type: text/plain; charset="utf-8"; Format="flowed"
> >
> > Hello everyone, I would like to show my part of the code when playing
> > early media after 200OK, when creating dialogs, I substituted $DLG_did
> > in the contact of my dialog, and received the same $DLG_did for my
> > dialog in ACK, but OpenSIPS also continued to send 200OK , despite
> > having already received an ACK response.
> >
> > route {
> >
> > # initial invite
> >
> > if (is_method("INVITE")) {
> >
> > create_dialog();
> >
> > route(early_media);
> >
> > exit;
> >
> > }
> >
> > } route[early_media] { if (has_body("application/sdp")) {
> > rtpengine_manage(); } $json(reply) := $rtpquery;
> > $var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
> > remove_body_part();
> >
> > append_to_reply("Contact:
> >  $socket_in(ip):$socket_in(port);transport=udp;did=$DLG_did>\r\n");
> >
> > append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
> > $(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
> > $(var(body){re.subst,/(audio.)./\1$var(port)/g});
> > t_reply_with_body(200, "OK", $var(body));
> > rtpengine_play_media("call-id=$ci from-tag=$ft
> > file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); }
> > route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete();
> > exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit;
> > } }
> >
> > I don’t know if Kevin example was with creating a dialog, but I also
> > noticed this problem through transaction... thanks
> > -- next part --
> > An HTML attachment was scrubbed...
> > URL:<
> http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html
> >
> > --
> >
> > Message: 1
> > Date: Thu, 2 Nov 2023 16:32:02 -0700
> > From: Kevin Kennedy
> > To: OpenSIPS users mailling list
> > Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not
> >   absorbing ACK
> > Message-ID:
> >   <
> cabdxsrxltp2_uex_upx1adg16af6gaetzjujutpki8c7h3k...@mail.gmail.com>
> > Content-Type: text/plain; charset="utf-8"
> >
> > I am trying to build a solution where Opensips 3.2+ with RTPengine acts
> as
> > a UAC, answers a call with 200OK, plays media from file, and will
> terminate
> > the call right after playing announcement.
> >
> > Opensips is responding with 200OK with SDP body and making the
> > correct changes for the IP, but 

Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Dmitry Ponomaryov]

It turns out that this is no early_media, there were simply successful 
attempts with 183 Session Progress, which is why there was such a 
misunderstanding, I’ll attach the snippet code again in plain text: 
route { if (is_method("INVITE")) { create_dialog(); route(media); exit; 
} } route[media] { if (has_body("application/sdp")) { rtpengine_offer(); 
} $json(reply) := $rtpquery; 
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); 
remove_body_part(); append_to_reply("Contact: 
\r\n"); 
append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = 
$(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = 
$(var(body){re.subst,/(audio.)./\1$var(port)/g}); 
t_reply_with_body(200, "OK", $var(body)); 
rtpengine_play_media("call-id=$ci from-tag=$ft 
file=/etc/rtpengine/media.wav"); async(sleep(10), after_media); } 
route[after_media] { if (t_was_cancelled()) { rtpengine_delete(); exit; 
} else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; } }


and pined previous posts below :)


--
Message: 2
Date: Fri, 3 Nov 2023 16:00:22 +0500
From: Dmitry Ponomaryov
To:users@lists.opensips.org
Subject: Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not
absorbing ACK
Message-ID:<3971cbc2-7281-2299-4212-7f241e8b8...@gmail.com>
Content-Type: text/plain; charset="utf-8"; Format="flowed"

Hello everyone, I would like to show my part of the code when playing
early media after 200OK, when creating dialogs, I substituted $DLG_did
in the contact of my dialog, and received the same $DLG_did for my
dialog in ACK, but OpenSIPS also continued to send 200OK , despite
having already received an ACK response.

route {

# initial invite

if (is_method("INVITE")) {

create_dialog();

route(early_media);

exit;

}

} route[early_media] { if (has_body("application/sdp")) {
rtpengine_manage(); } $json(reply) := $rtpquery;
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port);
remove_body_part();

append_to_reply("Contact:
\r\n");

append_to_reply("Content-Type: application/sdp\r\n"); $var(body) =
$(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) =
$(var(body){re.subst,/(audio.)./\1$var(port)/g});
t_reply_with_body(200, "OK", $var(body));
rtpengine_play_media("call-id=$ci from-tag=$ft
file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); }
route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete();
exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit;
} }

I don’t know if Kevin example was with creating a dialog, but I also
noticed this problem through transaction... thanks
-- next part --
An HTML attachment was scrubbed...
URL:<http://lists.opensips.org/pipermail/users/attachments/20231103/059cb479/attachment-0001.html>
----------

Message: 1
Date: Thu, 2 Nov 2023 16:32:02 -0700
From: Kevin Kennedy
To: OpenSIPS users mailling list
Subject: [OpenSIPS-Users] Opensips and rtpengine_play_media not
absorbing ACK
Message-ID:

Content-Type: text/plain; charset="utf-8"

I am trying to build a solution where Opensips 3.2+ with RTPengine acts as
a UAC, answers a call with 200OK, plays media from file, and will terminate
the call right after playing announcement.

Opensips is responding with 200OK with SDP body and making the
correct changes for the IP, but when the ACK comes back from the UAS,
Opensips doesn't seem to absorb it and retransmits the 200OK.

Code snippet handling this scenario

 rtpengine_manage("from-tag=$ft replace-session-connection
trust-address replace-origin codec-strip-g729",,$var(body));
 append_to_reply("Contact:\r\n");
 append_to_reply("Content-Type: application/sdp\r\n");
 t_reply_with_body(200, "OK", $var(body));
 rtpengine_play_media("from-tag=$ft
file=/etc/rtpengine/unk_num.wav");
 sleep(10);
 rtpengine_delete("from-tag=$ft");
 #t_reply(603, "Decline");
 exit();


What do I need to add to handle this scenario correctly?

Note:  I was able to get this to work with Early Media (183
reply_with_body, and send t_reply(603, "Decline")), but we have customers
using late media invite as well, so the Early Media option wouldn't work in
that case.

Thank you.

Kevin Kennedy
-- next part --
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Re: [OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Dmitry Ponomaryov]

Hello everyone, I would like to show my part of the code when playing 
early media after 200OK, when creating dialogs, I substituted $DLG_did 
in the contact of my dialog, and received the same $DLG_did for my 
dialog in ACK, but OpenSIPS also continued to send 200OK , despite 
having already received an ACK response.


route {

# initial invite

if (is_method("INVITE")) {

create_dialog();

route(early_media);

exit;

}

} route[early_media] { if (has_body("application/sdp")) { 
rtpengine_manage(); } $json(reply) := $rtpquery; 
$var(port)=$json_pretty(reply/tags/$ft/medias[0]/streams[0]/local port); 
remove_body_part();


append_to_reply("Contact: 
\r\n");


append_to_reply("Content-Type: application/sdp\r\n"); $var(body) = 
$(rb{re.subst,/(IP4.).*/\1$socket_in(ip)/g}); $var(body) = 
$(var(body){re.subst,/(audio.)./\1$var(port)/g}); 
t_reply_with_body(200, "OK", $var(body)); 
rtpengine_play_media("call-id=$ci from-tag=$ft 
file=/etc/rtpengine/media.wav"); async(sleep(10), after_early_media); } 
route[after_early_media] { if (t_was_cancelled()) { rtpengine_delete(); 
exit; } else { rtpengine_delete(); sl_send_reply(486,"Busy here"); exit; 
} }


I don’t know if Kevin example was with creating a dialog, but I also 
noticed this problem through transaction... thanks
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[OpenSIPS-Users] Opensips and rtpengine_play_media not absorbing ACK

[Kevin Kennedy]

I am trying to build a solution where Opensips 3.2+ with RTPengine acts as
a UAC, answers a call with 200OK, plays media from file, and will terminate
the call right after playing announcement.

Opensips is responding with 200OK with SDP body and making the
correct changes for the IP, but when the ACK comes back from the UAS,
Opensips doesn't seem to absorb it and retransmits the 200OK.

Code snippet handling this scenario

rtpengine_manage("from-tag=$ft replace-session-connection
trust-address replace-origin codec-strip-g729",,$var(body));
append_to_reply("Contact: \r\n");
append_to_reply("Content-Type: application/sdp\r\n");
t_reply_with_body(200, "OK", $var(body));
rtpengine_play_media("from-tag=$ft
file=/etc/rtpengine/unk_num.wav");
sleep(10);
rtpengine_delete("from-tag=$ft");
#t_reply(603, "Decline");
exit();


What do I need to add to handle this scenario correctly?

Note:  I was able to get this to work with Early Media (183
reply_with_body, and send t_reply(603, "Decline")), but we have customers
using late media invite as well, so the Early Media option wouldn't work in
that case.

Thank you.

Kevin Kennedy
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[OpenSIPS-Users] OpenSIPS as websocket client

[Ihor Olkhovskyi]

Hello,
I'm a bit new (to a recent versions) to OpenSIPS and trying it to act as a
UDP - WebSocket proxy using it as an outbound proxy in SIP client (PJSUA,
if it's important)

Currently I'm using 3.4.2 version.
Config is quite simple, not far from default one.
...
socket=udp:0.0.0.0:6051
socket=wss:0.0.0.0:9443
...
loadmodule "proto_udp.so"
loadmodule "proto_tls.so"

# WebSocket part
loadmodule "proto_wss.so"

loadmodule "tls_openssl.so"
loadmodule "tls_mgm.so"

modparam("tls_mgm", "client_domain", "localhost")
modparam("tls_mgm", "certificate",
"[localhost]/etc/ssl/certs/ssl-cert-snakeoil.pem")
modparam("tls_mgm", "private_key",
"[localhost]/etc/ssl/private/ssl-cert-snakeoil.key")
modparam("tls_mgm", "ca_list",
"[localhost]/etc/ssl/certs/ca-certificates.crt")
modparam("tls_mgm", "verify_cert", "[localhost]0")
modparam("tls_mgm", "require_cert", "[localhost]0")

...
route[relay] {
if ($socket_in(proto) == "UDP") {
$socket_out = "wss:0.0.0.0:9443";
} else {
$socket_out = "udp:0.0.0.0:6051";
}

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

I'm using most generic self-signed certs and just started to make some
experiments.
But when I'm trying just forward SIP packets to remote server, I'm getting
this in the logs

DBG:core:parse_headers: flags=
DBG:proto_wss:proto_wss_send: no open tcp connection found, opening new one
DBG:core:probe_max_sock_buff: getsockopt: snd is initially 16384
DBG:core:probe_max_sock_buff: using snd buffer of 416 kb
DBG:core:init_sock_keepalive: TCP keepalive enabled on socket 4
DBG:core:print_ip: tcpconn_new: new tcp connection to: 
DBG:core:tcpconn_new: on port 8089, proto 6
DBG:tls_mgm:tls_find_client_domain: found TLS client domain: localhost
DBG:tls_openssl:openssl_tls_conn_init: Creating a whole new ssl connection
DBG:tls_openssl:openssl_tls_conn_init: Setting in CONNECT mode (client)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
ERROR:tls_openssl:openssl_tls_blocking_write: TLS send timeout (100)
ERROR:proto_wss:ws_client_handshake: cannot start handshake
ERROR:proto_wss:ws_connect: cannot complete WebSocket handshake
DBG:core:tcpconn_destroy: destroying connection 0x7f0efb106440, flags 0038
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
NOTICE:tls_openssl:verify_callback: depth = 2, verify success
NOTICE:tls_openssl:verify_callback: depth = 1, verify success
NOTICE:tls_openssl:verify_callback: depth = 0, verify success
INFO:tls_openssl:openssl_tls_connect: New TLS connection to
:8089 established
DBG:tls_openssl:openssl_tls_connect: new TLS connection to
:8089 using TLSv1.2 ECDHE-RSA-AES256-GCM-SHA384 256
DBG:tls_openssl:openssl_tls_connect: sending socket: 0.0.0.0:37697
INFO:tls_openssl:tls_dump_cert_info: tls_connect: server TLS certificate
subject: /CN=*.pbx.company.domain, issuer: /C=GB/ST=Greater
Manchester/L=Salford/O=Sectigo Limited/CN=Sectigo RSA Domain Validation
Secure Server CA
INFO:tls_openssl:tls_dump_cert_info: tls_connect: local TLS client
certificate subject: /CN=localhost, issuer: /CN=localhost
DBG:tls_openssl:openssl_tls_write: write was successful (6 bytes)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
DBG:tls_openssl:openssl_tls_write: write was successful (2 bytes)
DBG:tls_openssl:openssl_tls_update_fd: New fd is 4
DBG:tls_openssl:openssl_tls_conn_shutdown: first phase of 2-way handshake
completed succesfuly
ERROR:proto_wss:proto_wss_send: connect failed
ERROR:tm:msg_send: send() to :8089 for proto wss/6 failed
ERROR:tm:t_forward_nonack: sending request failed
DBG:tm:t_relay_to: t_forward_nonack returned error


Server that I'm making connections to is supporting TLS and WSS transports.
If I'm changing socket type from WSS to TLS, all is working, so it's not a
TLS certificate issue or something like this.

I'm pretty sure, that I'm missing something obvious, but not really getting
what.

Would be appreciated for any hints.
-- 
Best regards,
Ihor (Igor)
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Re: [OpenSIPS-Users] Opensips on vmware and multiple CPUs

[Bogdan-Andrei Iancu]

Hi Simon,

You can try to run an "opensipsctl trap" to see what are the opensips 
processes doing - just to be sure opensips as application is actually 
listening and ready to handle packages.


Best regards,

Bogdan-Andrei Iancu

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

On 10/13/23 11:43 AM, Simon Gajski via Users wrote:

Hi

I am running one older instance of OpenSIPS (2.1.5 (x86_64/linux)) on 
VMware 1CPU/1core
Lately we have some performance issues with the server so I tried to 
increase number of  CPU to 2 and 2 cores per CPU


In opensips script is set to worker=4
Opensips starts normally after change on vmware, however it doesn't 
respond to any SIP request anymore.
Once I set back to 1CPU/1core it starts responding back to all SIP 
requests.


Not sure where to start digging; vmware, linux or opensips?
Gratefull for any tip

Thank you

BR
Simon



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