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Summary:ASTERISK-26552: chan_sip: chan_pjsip to chan_sip call, CANCEL received on pjsip, CANCEL generated on chan_sip way too late
Reporter:Daniel Heckl (DanielYK)Labels:
Date Opened:2016-11-03 04:27:40Date Closed:
Priority:MajorRegression?
Status:Open/NewComponents:Channels/chan_sip/General
Versions:13.12.1 Frequency of
Occurrence
Related
Issues:
Environment:Asterisk 13.12.1 bundled with the newest pjsip version (2.5.5)Attachments:( 0) pjsip.conf
( 1) sip_pjsip.txt
( 2) sip.conf
Description:We use for our local phones pjsip and sip for our trunks.

If the pjsip phones just hangup while the sip channel is ringing, sometimes the hangup of the sip channel is not performed.

I have attached a log with this situation.

Asterisk receives at 10:06:27 a CANCEL from the pjsip phone. The debugging log shows
[2016-11-03 10:06:28] DEBUG[28966][C-00000056]: chan_sip.c:7145 sip_hangup: Hangup call SIP/061129634568-00000045, SIP callid 43fc1d2572c7738c02a4c773605154fe@tel.t-online.de
but a CANCEL is not send. At 10:06:53 a CANCEL is send to the trunk, 25 seconds too late.
Comments:By: Asterisk Team (asteriskteam) 2016-11-03 04:27:41.693-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Rusty Newton (rnewton) 2016-11-03 09:13:00.310-0500

Daniel, thanks for the report.

Please additionally attach

* sip.conf
* pjsip.conf
* Asterisk version
* PJSIP version and whether you used bundled or not

As an aside, why are you using chan_sip while using chan_pjsip?  chan_sip has been moved to extended support and the development focus is on the core supported pjsip driver now.
By: Daniel Heckl (DanielYK) 2016-11-03 17:01:35.893-0500

Attached all needed config files. We use Asterisk 13.12.1 bundled with the newest pjsip version (2.5.5).

We have to use sip and pjsip both, because pjsip is not ready for use with dynamic public ip addresses (external_media_address has no domain support). We would prefer to use only pjsip, but we have to wait until pjsip integration is better.
By: Rusty Newton (rnewton) 2016-11-10 08:01:51.671-0600

Thanks I'm opening the issue up. Remember that chan_sip has moved from core support to extended support, so it is up to the broader community as a whole to investigate and resolve this issue. That may be reflected in the time it takes to address the issue.