8 Nov
2004
8 Nov
'04
9:35 p.m.
Hi Graziano,
most probably your routing is different fro INVITEs and CANCELs - check
if you don't have in your script some specific only for INVITEs and not
for CANCEL.
CANCEL must be routed all the time on the same criteria as INVITE.
Best regards,
Marian
Graziano Poretti wrote:
hi marianu
it works .. thnx. the only problem now is that if i end the conversation
from my ip phone before that on the other side will reply, on the other
side doesn't stop ringing ... is this correct or it's a fault (may be i
have to change something on my ser/asterisk)
sorry .. i'm a newbie here ... may be that this was a silly question,
but thnx in any case.
thnx
RAINBOW TEAM Graziano Poretti Rainbow Product Manager Due Metri s.a.s.
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----- Original Message ----- From: "Marian Dumitru"
<marian.dumitru(a)voice-sistem.ro>
To: "Graziano Poretti" <graziano(a)duemetri.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Saturday, November 06, 2004 11:31 AM
Subject: Re: [Serusers] ser+asterisk again
--
Voice Sistem
http://www.voice-sistem.ro
Hi Graziano
if you want to divert the calls with "00" from ser to a asterisk, you
just need to add the following lines in your ser.cfg (probably
somewhere inside "uri==myself" if consition, just before
lookup("location") - if you are using standard script) :
if (uri=~"sip:00") {
t_relay_to_udp("asterisk_IP","asterisk_port");
break;
}
Best regards,
Marian Dumitru
Graziano Poretti wrote:
hi
we have 1 ser + 1 asterisk on different computers
we want to forward incoming calls from the server with fixed prefix
(eg 00) to asterisk.
i tried different solutions but how is the connection logic between
ser and asterisk? should i have a registered user in asterisk which
collects all the traffic coming from ser or not? how can i setup the
ser and asterisk to have then "talking"?
thnx in advance for help
Graziano
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