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

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-22870?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-22870:
------------------------------------

    Assignee: Arno Teigseth
      Status: Waiting for Feedback  (was: Triage)

It is probably hung up in DNS resolution. A quick little test on my system showed that Asterisk does attempt to resolve the non-existing peer as a hostname. Although I don't get a hang of any kind.

{noformat}

    -- Executing [6009 at from-internal:1] Dial("SIP/6001-00000002", "SIP/barneyfief&SIP/6002,10") in new stack

<snip>

[Nov 21 15:18:42] DEBUG[21380][C-00000001]: netsock2.c:138 ast_sockaddr_split_hostport: Splitting 'barneyfief' into...
[Nov 21 15:18:42] DEBUG[21380][C-00000001]: netsock2.c:192 ast_sockaddr_split_hostport: ...host 'barneyfief' and port ''.
[Nov 21 15:18:42] ERROR[21380][C-00000001]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("barneyfief", "(null)", ...): No address associated with hostname
[Nov 21 15:18:42] WARNING[21380][C-00000001]: chan_sip.c:6201 create_addr: No such host: barneyfief
{noformat}

Do another test with DEBUG messages turned up to 5 in addition to your VERBOSE messages. (see logger.conf and check "logger show channels" on the CLI) You may find the issue there, but if not, attach the results to the issue.


                
> 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
>            Assignee: 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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