[OpenSIPS-Users] Dialog destroys when answering call!!

[Richard Revels]

Sorry.  Should have wondered what the avp is set to when the dialog is created rather than when it is destroyed.


On May 10, 2010, at 12:33 PM, Richard Revels wrote:

> I wonder what the timeout_avp is set to when the dialog goes away?  Not to get off topic, but if anyone knows, can that avp be modified via opensipsctl?  Been meaning to investigate this question and keep forgetting.
> 
> Richard
> 
> 
> On May 10, 2010, at 9:17 AM, Neo Anderson wrote:
> 
>> Hello Bogdan,
>> 
>> Yes when opensips gets 200 OK & then send ACK back to the Callee, that time dialog destroys.
>> When making call, I am executing opensipsctl fifo dlg_list .
>> Till receiving 200 OK I can see the dialog but when sending ACK dialog destroys. Also I found Status 5 in dialog list.
>> Please let me know if anything I did wrong in configurations.
>> 
>> dialog module configurations:
>> modparam("dialog", "dlg_flag", 10)
>> modparam("dialog", "dlg_match_mode", 1)
>> modparam("dialog", "profiles_with_value", "caller")
>> modparam("dialog", "default_timeout", 43200)                                
>> modparam("dialog", "timeout_avp", "$avp(i:100)")
>> 
>> Thanks.
>> 
>> --
>> Neo
>> 
>> 
>> 
>> 
>> From: Bogdan-Andrei Iancu <bogdan at voice-system.ro>
>> To: OpenSIPS users mailling list <users at lists.opensips.org>
>> Sent: Mon, May 10, 2010 6:20:55 PM
>> Subject: Re: [OpenSIPS-Users] Dialog destroys when answering call!!
>> 
>> Hi Neo,
>> 
>> you are saying the dialogs (in dialog module) are destroyed when they 
>> are answered (200 ok ) ? what makes you say that? I mean what do you see 
>> to confirm this?
>> 
>> Regards,
>> Bogdan
>> 
>> Neo Anderson wrote:
>> > Hi,
>> >
>> > I am using OpenSIPS 1.5.3 .
>> > I have implemented call-limit based on the tutorial.
>> >
>> > http://www.opensips.org/Resources/DocsTutConcurrentCalls 
>> > <http://www.opensips.org/Resources/DocsTutConcurrentCalls>
>> >
>> > But when call gets answered, dialog destroys. That's why call limit is 
>> > not working.
>> > Would you please let me know what I am doing wrong?
>> > I have followed the same instructions given in the tutorial.
>> > I am using carrier-route module to route outbound calls.
>> >
>> > Thanks in advance!!!
>> >
>> > --
>> > Neo
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
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>> > Users at lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >  
>> 
>> 
>> -- 
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>> 
>> 
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