[Serusers] Dynamic Invite Timer w/AVPOPS
Marian Dumitru
marian.dumitru at voice-sistem.ro
Fri Dec 3 19:54:55 CET 2004
Hi Paul,
The script looks ok. The problem is in other part; it took my some time
to trace it - there is a bug in TM in setting the "fr_inv_timer_avp"
parameter. Because of this, the AVP was not found and default value was
applied for timeout.
Please find attach a patch to fix the problem.
Best regards,
Marian
Java Rockx wrote:
> Thanks, however I must be doing something wrong because the avp_write() doesn't seem to have any
> effect on the INVITE timer timeout.
>
> My ser.cfg looks like this:
>
> # ------------- tm parameters
> modparam("tm", "fr_timer", 15)
> modparam("tm", "fr_inv_timer", 22)
> modparam("tm", "wt_timer", 5)
> modparam("tm", "fr_inv_timer_avp", "inv_timeout")
>
> route {
>
> # sanity checks, record route, alias lookup, etc, etc
>
> if (!lookup("location")) {
>
> if (uri=~"^sip:[0-9]{10}@") {
>
>
> # Send to PSTN Gateway
> avp_write("45", "inv_timeout");
> route(3);
> break;
> }
>
> sl_send_reply("404", "User Not Found");
> break;
> };
>
> # normal SIP->SIP routing with default fr_inv_timer
> # set to 22 seconds with mod_param() above
> route(2);
> }
>
> Shouldn't this do the trick?
>
> Regards,
> Paul
>
>
> --- Marian Dumitru <marian.dumitru at voice-sistem.ro> wrote:
>
>
>>Hi Paul,
>>
>>you can control final_response_timer and final_response_invite_timer via
>> avps. Set in TM fr_inv_avp / fr_inv_timer_avp parameters with some AVP
>>string names and use AVPOPS to give values to these AVPS:
>>
>>modparam("tm","fr_inv_timer_avp","inv_timeout")
>>
>>....
>>avp_write("20","inv_timeout"); # set timeout to 20 sec
>>t_relay();
>>.....
>>
>>Best regards,
>>Marian
>>
>>Java Rockx wrote:
>>
>>>Hi All.
>>>
>>>I thought I read something a while ago about using avpops to dymanically change the INVITE
>>
>>timer -
>>
>>>but I can't seem to find that thread.
>>>
>>>Can anyone tell me if this is possible?
>>>
>>>I'd like to have a shorter INVITE timer for SIP->SIP calls and a longer INVITE timer for
>>
>>SIP->PSTN
>>
>>>calls. We see some 408 replies because a SIP phone is dialing a cell phone and get transfered
>>
>>to
>>
>>>the voicemail of the cell phone user, but to prevent the 408 due to a slow voicemail system we
>>>have our INVITE timer currently set at 40 seconds which is rather annoying for SIP->SIP calls.
>>>
>>>I'm using -dev-21
>>>
>>>Regards,
>>>Paul
>>>
>>>
>>>
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>>
>>--
>>Voice Sistem
>>http://www.voice-sistem.ro
>>
>
>
>
>
>
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