[asterisk-bugs] [JIRA] (ASTERISK-24846) Cancel Request Broken in chan_pjsip when it's used on Trunk with TCP transport

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24846?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-24846:
-----------------------------------

    Assignee: Javier Riveros 
      Status: Waiting for Feedback  (was: Triage)

It's not. There are tests in the test suite which covers its use. In your specific situation your configuration is such that no existing connection will be used. Remove all "transport" lines from endpoint configurations and retest.

> Cancel Request Broken in chan_pjsip when it's used on Trunk with  TCP transport
> -------------------------------------------------------------------------------
>
>                 Key: ASTERISK-24846
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24846
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 12.8.1, 13.2.0
>         Environment: Asterisk 13.2
> OS : Ubuntu 14.04
> Standart softphone 
>            Reporter: Javier Riveros 
>            Assignee: Javier Riveros 
>         Attachments: Full_Ast_Debug_Cancel_TCP_trunk_pjsip, pjsip.conf.txt, Sip_capture_cancel.txt
>
>
> When asterisk request a CANCEL on a trunk it use different origin port used on INVITE it cause the trunk endpoint dont hangup the call because transaction doesn't exist.
> {code}
> - INVITE Generate by asterisk
> T 172.31.22.69:"56625" -> 172.31.16.57:5060 [A]
> INVITE sip:pbx.rarevase.com;transport=tcp SIP/2.0.
> Via: SIP/2.0/TCP 172.31.22.69:56625;rport;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>.
> Contact: <sip:425b1b25-1ed5-49d6-a561-52ed63c566dd at 172.31.22.69:56625>.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> -CANCEL Send by asterisk
> ###
> T 172.31.22.69:"51875" -> 172.31.16.57:5060 [AP]
> CANCEL sip:pbx.rarevase.com;transport=tcp SIP/2.0.
> Via: SIP/2.0/TCP 172.31.22.69:51875;rport;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> -- Response from the gateway trunk
> T 172.31.16.57:5060 -> 172.31.22.69:"51875" [AP]
> SIP/2.0 481 Call/Transaction Does Not Exist.
> Via: SIP/2.0/TCP 172.31.22.69:51875;rport=51875;received=172.31.22.69;branch=z9hG4bKPj70f06330-08c0-48ae-8588-c3ab27839c68;alias.
> From: 'javier' <sip:javier at 172.31.22.69>;tag=a15f98b6-5f03-4b55-84f1-5f677759711a.
> To: <sip:c4e1a3 at pbx.rarevase.com>;tag=awAdaH.
> Call-ID: 2aae6f80-f2c5-4c9b-94a0-37cfe99d2485.
> {code} 
> If you see you can check that Call-ID is the same for all outbound transaction but the origin port on asterisk site is {{51875}} used for CANCEL is different it was using initially on INVITE {{56625}} so this causes that GATEWAY doesn't hangup the call in the other party.
> To reproduce this use a trunk using TCP transport  on chan_pjsip.
> i attached the full asterisk debug, sip_capture and pjsip.conf config. 



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