[asterisk-bugs] [JIRA] (ASTERISK-25799) can't place calls to throught sip trunk to cisco

Asterisk Team (JIRA) noreply at issues.asterisk.org
Wed Feb 17 10:41:33 CST 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-25799?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=229551#comment-229551 ] 

Asterisk Team commented on ASTERISK-25799:
------------------------------------------

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].

> can't place calls to throught sip trunk to cisco
> ------------------------------------------------
>
>                 Key: ASTERISK-25799
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25799
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Addons/General
>    Affects Versions: 1.8.25.0
>            Reporter: Rachid HIGUI
>
> I have set up a sip trunk between Asterisk (192.5.0.207) and Cisco Cucm 7.1 (192.5.0.201)
> Extensions can call each other.
> Cisco ip phones can call Asterisk softphones.
> The problem is that Asterisk softphones can't call Cisco ip phones
> Below is a sammury of the log:
> [Feb 17 08:08:31] VERBOSE[22054] app_dial.c:     -- SIP/trunk_1-0000001d is circuit-busy
> [Feb 17 08:08:31] VERBOSE[22054] app_dial.c:   == Everyone is busy/congested at this time (1:0/1/0)
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Executing [s at macro-trunkdial-failover:35] Set("SIP/800-0000001c", "num=4") in new stack
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Executing [s at macro-trunkdial-failover:36] GotoIf("SIP/800-0000001c", "0>0?s-CONGESTION,1:s-out,1") in new stack
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Goto (macro-trunkdial-failover,s-out,1)
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Executing [s-out at macro-trunkdial-failover:1] ExecIf("SIP/800-0000001c", "0?System(/etc/scripts/faxlog.sh    "FAILED" "CONGESTION")") in new stack
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Executing [s-out at macro-trunkdial-failover:2] StopMixMonitor("SIP/800-0000001c", "") in new stack
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:     -- Executing [s-out at macro-trunkdial-failover:3] Congestion("SIP/800-0000001c", "10") in new stack
> [Feb 17 08:08:31] VERBOSE[22054] app_macro.c:   == Spawn extension (macro-trunkdial-failover, s-out, 3) exited non-zero on 'SIP/800-0000001c' in macro 'trunkdial-failover'
> [Feb 17 08:08:31] VERBOSE[22054] pbx.c:   == Spawn extension (DLPN_DialPlan1, 486, 1) exited non-zero on 'SIP/800-0000001c'



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