asterisk users - Feb 2016 - Unexpected termination of the call when pick up (res_pjsip)

The call initiated from internal extension.

I have made two test call:
Successful call from device on res_pjsip via endpoint on chan_sip: 
http://pastebin.com/LWeDYstj
Unsuccessful call from device on res_pjsip via endpoint on res_pjsip: 
http://pastebin.com/hepVb6Nu

And ones again i don't see anything that would make asterisk send BYE.

I would be grateful for any ideas.

11.02.2016 1:47, Trey Hilyard ?????:
>
> How are you initiating the call out to that server? Are you dialing 
> from an internal phone or doing it from the CLI? It looks like it is 
> from an internal extension, if I were guessing, but that side of the 
> call isn't in your log.
>
> If it is from an internal extension, I think a SIP trace on that side 
> would help.
>
>
> On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov.d.p at gmail.com 
> <mailto:serov.d.p at gmail.com>> wrote:
>
>     Please help find the cause of strange behavior res_pjsip.
>
>     Making outgoint call to other sip server (CommuniGatePro), my
>     asterisk suddenly sends BYE after picking up!
>     Partial log of an outgoing call with full debug is attached and on
>     web: http://pastebin.com/tLNCpx4d
>
>     No diagnostic messages why asterisk suddenly decided to hangup i
>     don't found :(
>
>     There are suggestions or strong belief about the reasons of such
>     behavior?
>
>     Thanks.
>
>     Dmitriy.
>     --
>     _____________________________________________________________________
>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>     New to Asterisk? Join us for a live introductory webinar every Thurs:
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>
>
>
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I am stumped so far. What is most interesting to me is that Asterisk is
actually sending two BYE transactions for the same dialog, at basically the
same time. I am still going through your traces again, but maybe someone
else has suggestions on how to add more debug to the res_pjsip logging that
would prove useful.

On Thu, Feb 11, 2016 at 1:33 AM Dmitriy Serov <serov.d.p at gmail.com>
wrote:
> The call initiated from internal extension.
>
> I have made two test call:
> Successful call from device on res_pjsip via endpoint on chan_sip:
> http://pastebin.com/LWeDYstj
> Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
> http://pastebin.com/hepVb6Nu
>
> And ones again i don't see anything that would make asterisk send BYE.
>
> I would be grateful for any ideas.
>
> 11.02.2016 1:47, Trey Hilyard ?????:
>
> How are you initiating the call out to that server? Are you dialing from
> an internal phone or doing it from the CLI? It looks like it is from an
> internal extension, if I were guessing, but that side of the call isn't
in
> your log.
>
> If it is from an internal extension, I think a SIP trace on that side
> would help.
>
> On Wed, Feb 10, 2016, 3:20 PM Dmitriy Serov <serov.d.p at gmail.com>
wrote:
>
>> Please help find the cause of strange behavior res_pjsip.
>>
>> Making outgoint call to other sip server (CommuniGatePro), my asterisk
>> suddenly sends BYE after picking up!
>> Partial log of an outgoing call with full debug is attached and on web:
>> http://pastebin.com/tLNCpx4d
>>
>> No diagnostic messages why asterisk suddenly decided to hangup i
don't
>> found :(
>>
>> There are suggestions or strong belief about the reasons of such
behavior?
>>
>> Thanks.
>>
>> Dmitriy.
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
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Dmitriy Serov wrote:
> The call initiated from internal extension.
>
> I have made two test call:
> Successful call from device on res_pjsip via endpoint on chan_sip:
> http://pastebin.com/LWeDYstj
> Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
> http://pastebin.com/hepVb6Nu
>
> And ones again i don't see anything that would make asterisk send BYE.
>
> I would be grateful for any ideas.
Kia ora,

I have a feeling it may be an off-nominal SDP negotiation issue, causing 
two paths to get triggered which both send a BYE. I'd suggest filing an 
issue[1] with the traces you've provided. We can potentially use them to 
construct a sipp scenario that reproduces the issue. The configuration 
would also be needed.

You can also try to narrow it down slightly by disabling the video 
codecs and seeing if that changes things. If it does then it's with 
video involved.

[1] https://issues.asterisk.org/jira

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

11.02.2016 18:50, Joshua Colp ?????:
> Dmitriy Serov wrote:
>> The call initiated from internal extension.
>>
>> I have made two test call:
>> Successful call from device on res_pjsip via endpoint on chan_sip:
>> http://pastebin.com/LWeDYstj
>> Unsuccessful call from device on res_pjsip via endpoint on res_pjsip:
>> http://pastebin.com/hepVb6Nu
>>
>> And ones again i don't see anything that would make asterisk send
BYE.
>>
>> I would be grateful for any ideas.
>
> Kia ora,
>
> I have a feeling it may be an off-nominal SDP negotiation issue, 
> causing two paths to get triggered which both send a BYE. I'd suggest 
> filing an issue[1] with the traces you've provided. We can potentially 
> use them to construct a sipp scenario that reproduces the issue. The 
> configuration would also be needed.
>
> You can also try to narrow it down slightly by disabling the video 
> codecs and seeing if that changes things. If it does then it's with 
> video involved.
>
> [1] https://issues.asterisk.org/jira
>
Joshua, Thanks! Disabling all codecs except alaw (I guess video codecs) 
makes call successful.

https://issues.asterisk.org/jira/browse/ASTERISK-25772