[asterisk-users] Unexpected termination of the call when pick up (res_pjsip)

Trey Hilyard kctrey at gmail.com
Thu Feb 11 09:26:05 CST 2016


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 пишет:
>
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160211/22337223/attachment.html>


More information about the asterisk-users mailing list