[asterisk-bugs] [JIRA] (ASTERISK-22870) dialplan entries pointing to SIP peers not defined in sip.conf just hangs the call

Arno Teigseth (JIRA) noreply at issues.asterisk.org
Wed Nov 20 08:46:04 CST 2013


    [ https://issues.asterisk.org/jira/browse/ASTERISK-22870?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=212011#comment-212011 ] 

Arno Teigseth commented on ASTERISK-22870:
------------------------------------------

Tried also to see if the pbx tries to resolve SIP/arno to a hostname, and calling that hostname. Sniffing the network leads me to suspect that isn't the case, since I see no outgoing requests for hostname "arno".

And adding a hostname "arno" to /etc/hosts yields no connection attepmts to the host.

Seems to me as asterisk just gives up on finding SIP peer arno when it isn't defined in sip.conf, and abandons the call without hanging up, error code or anything.
                
> dialplan entries pointing to SIP peers not defined in sip.conf just hangs the call
> ----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-22870
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-22870
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_dial
>    Affects Versions: 11.3.0
>         Environment: alpine linux 2.5 edge
>            Reporter: Arno Teigseth
>            Severity: Minor
>         Attachments: nonexistantSipbugfeature.txt
>
>
> I had set up in extensions.conf
> exten => 9904,1,Dial(SIP/930&SIP/arno)
> Usually when an extension isn't registered, it's just ignored when dialling 9904.
> Now, I deleted SIP peer "arno" from sip.conf and now dialling 9904 it just hangs. No audio, no ring on SIP peer 930 even if it is registered.
> To make things worse, if calling 9904 from an IAX2 trunk, the trunk hangs (see CLI verbosity 9 output attached). Only way to clear the condition is restarting the asterisk service. Which is bad.
> Of course, I should keep my pbx organized, so I don't know if this is a bug or a feature. But if there aren't negative side effects, wouldn't it be good if non-existant SIP peers were treated as non-registered ones?

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