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@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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