[asterisk-bugs] [JIRA] (ASTERISK-24848) INVITE not send by asterisk on chan_pjsip when endpoint has TCP Transport

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24848?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-24848:
-----------------------------------

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

Same here, except since it's an endpoint that is dynamic remove the transport= line and set rewrite_contact=yes.

This will cause the established TCP connection to get used instead of creating a new one.

> INVITE not send by asterisk on chan_pjsip when endpoint has TCP Transport
> -------------------------------------------------------------------------
>
>                 Key: ASTERISK-24848
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24848
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 12.8.1, 13.0.0, 13.1.1, 13.2.0
>         Environment: Asterisk 13.2 
> OS : Ubuntu 14.04 
> Standart softphone
>            Reporter: Javier Riveros 
>            Assignee: Javier Riveros 
>         Attachments: Full_Ast_Debug_INVITE_TCP.txt, pjsip_conf.txt, sip_capture.txt
>
>
> Asterisk not send INVITE to TCP transport endpoint , captures reflect that and also after some seconds it raise the following error.
> {code}
> [Mar  4 21:48:25] ERROR[14964]: pjsip:0 <?>: 	  tcpc0xb48588 TCP connect() error: Connection timed out [code=120110]
> {code}
> The result of this is that you can't make a SIP call to an endpoint that has TCP transport enable on chan_pjsip.
>  
> To reproduce this use a one or two SIP standart client and set this up with TCP transport.
> Full Debug , sip capture and pjsip_conf is attached



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