[asterisk-bugs] [JIRA] (ASTERISK-25688) Problem calling to a pjsip trunk

Joshua Colp (JIRA) noreply at issues.asterisk.org
Tue Jan 12 19:18:33 CST 2016


     [ https://issues.asterisk.org/jira/browse/ASTERISK-25688?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-25688:
-----------------------------------

    Severity: Minor  (was: Major)

> Problem calling to a pjsip trunk
> --------------------------------
>
>                 Key: ASTERISK-25688
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25688
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Core/BuildSystem
>    Affects Versions: 13.6.0
>         Environment: Asterisk 13.6.0 (tested in Asterisk 13.7.0 rc2 too), Ubuntu 14.04.3 LTS, Linux rtc 4.1.5-x86_64-linode61 #7 SMP Mon Aug 24 13:46:31 EDT 2015 x86_64 x86_64 x86_64 GNU/Linux
>            Reporter: Marcelo Terres
>            Assignee: Unassigned
>            Severity: Minor
>
> When I try to call to another Asterisk via pjsip trunk, Asterisk crashs:
>     -- Executing [407 at ramais:2] Progress("PJSIP/10-00000000", "") in new stack
>     -- Executing [407 at ramais:3] Answer("PJSIP/10-00000000", "") in new stack
>     -- Executing [407 at ramais:4] Dial("PJSIP/10-00000000", "PJSIP/407 at asterisk,30") in new stack
>     -- Called PJSIP/407 at asterisk
>        > 0x7f166001e6b0 -- Probation passed - setting RTP source address to 177.27.46.125:4000
>     -- PJSIP/10-00000000 requested media update control 26, passing it to PJSIP/asterisk-00000001
>        > 0x7f166001e6b0 -- Probation passed - setting RTP source address to 177.27.46.125:4000
>     -- PJSIP/asterisk-00000001 answered PJSIP/10-00000000
>     -- Channel PJSIP/asterisk-00000001 joined 'simple_bridge' basic-bridge <ee50b922-2431-4571-9627-33a3a6d43921>
>     -- Channel PJSIP/10-00000000 joined 'simple_bridge' basic-bridge <ee50b922-2431-4571-9627-33a3a6d43921>
>        > Bridge ee50b922-2431-4571-9627-33a3a6d43921: switching from simple_bridge technology to native_rtp
>        > Remotely bridged 'PJSIP/10-00000000' and 'PJSIP/asterisk-00000001' - media will flow directly between them
>        > Remotely bridged 'PJSIP/10-00000000' and 'PJSIP/asterisk-00000001' - media will flow directly between them
> asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_pjsip_session.so: undefined symbol: pj_timer_entry_running
> It is the same problem reported in this URL: http://stackoverflow.com/questions/34287633/asterisk-symbol-lookup-error-res-pjsip-session-so-undefined-symbol-pj-timer-ent



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