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 <bog...@voice-system.ro>
>> To: OpenSIPS users mailling list <users@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
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > Users mailing list
>> > Users@lists.opensips.org
>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>> >
>>
>>
>> --
>> Bogdan-Andrei Iancu
>> www.voice-system.ro
>>
>>
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