Re: [Serusers] ser+asterisk again - sr-users

8 Nov 2004

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. 
 graziano(a)duemetri.com tel: ++39 0184 42163 Signature powered by Plaxo 
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 ----- Original Message ----- From: "Marian Dumitru" 
 &lt;marian.dumitru(a)voice-sistem.ro&gt;
 To: "Graziano Poretti" &lt;graziano(a)duemetri.com&gt;
 Cc: &lt;serusers(a)lists.iptel.org&gt;
 Sent: Saturday, November 06, 2004 11:31 AM
 Subject: Re: [Serusers] ser+asterisk again

  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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