[SR-Users] if (t_check_status("486|408"))

hiro 23hiro at gmail.com
Sun Jun 2 22:57:30 CEST 2013


I'm still thinking about this issue and wondering:
is it even compliant to the RFC to go directly from ringing to session
progress and then OK? Because that's what freeswitch is answering with
when I try to relay the call to it's voicemail when user is busy in
kamailio.

On 6/1/13, hiro <23hiro at gmail.com> wrote:
> I tried for multiple hours to operate the debugger, also looking at
> -ddd output at stdout for many days.
> But I'm none the wiser.
> voicemail works from route[location] (e.g. if extension is not
> registered), but not after late errors like busy while ringing.
>
> On 5/23/13, Daniel-Constantin Mierla <miconda at gmail.com> wrote:
>>
>> On 5/19/13 2:05 PM, hiro wrote:
>>> i'm trying to use the example kamailio.cfg to route to voicemail
>>> server on busy or decline.
>>> Only thing I did was adding decline code to t_check_status("486|408"),
>>> enabling the preprocessor variable for voicemail and changing the
>>> voicemail host and port to my voicemail server.
>>> No requests arrive on my voicemail server and the dial tone keeps
>>> ringing even if phone is busy and when I decline on the receiving
>>> phone there's a new INVITE sent to the phone directly afterwards.
>>> Am I doing something wrong here?
>> enable debugger module with cfgtrace option and see if the right actions
>> are executed in the configuration file. Maybe there is a
>> misconfiguration or wrong condition somewhere.
>>
>> Cheers,
>> Daniel
>>
>> --
>> Daniel-Constantin Mierla - http://www.asipto.com
>> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
>>    * http://asipto.com/u/katu *
>>
>>
>> _______________________________________________
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>> sr-users at lists.sip-router.org
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>>
>



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