3 Dec
2004
3 Dec
'04
9:54 p.m.
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(a)voice-sistem.ro> wrote:
--
Voice Sistem
http://www.voice-sistem.ro
Index: modules/tm/t_funcs.c
===================================================================
RCS file: /cvsroot/ser/sip_router/modules/tm/t_funcs.c,v
retrieving revision 1.172
diff -u -r1.172 t_funcs.c
--- modules/tm/t_funcs.c 1 Nov 2004 14:09:08 -0000 1.172
+++ modules/tm/t_funcs.c 3 Dec 2004 18:49:14 -0000
@@ -340,7 +340,7 @@
fr_inv_timer_avp.s = &fr_inv_timer_param;
fr_timer_param.len = strlen(fr_timer_param.s);
- fr_inv_timer_param.len = strlen(fr_timer_param.s);
+ fr_inv_timer_param.len = strlen(fr_inv_timer_param.s);
}
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:
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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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