18 Jul
2007
18 Jul
'07
9:25 a.m.
Hi Jai,
Thanks for your answer.
It seems to have something like a loop. When I do the call, SER loop
between him and Asterisk.
Maybe Asterisk doesn't match the call, or the loop is generate by SER.
If somebody has experience in this kind of application :) I think it's
like a trunk.
Le mardi 17 juillet 2007 à 09:44 -0700, Jai Rangi a écrit :
If its an extension then asterisk must have the
extension. Otherwise
it will be treated like a did on asterisk, and in your dial plan you
can define something like this.
exten => enum,hint,SIP/yourextensionhere
This will ring yourextension when the call come for enum. Ofcourse you
need to make sure that this is called in proper context.
On ser you can check
if (uri=~"^enum(a)dimain.tld") {
rewritehost("asteriskip") ; //something like this. check the
syntax.
t_relay();
break;
};
Hope this helps,
On 7/17/07, inge <inge(a)legos.fr> wrote:
Hi all,
Anyone know how can I transfer an incoming call from SER to an
Asterisk ?
The sip uri wich comes from SER is like : sip:enum@domain.tld
But on Asterisk enum will not be necessary the extension.
IT seems that with a single rewritehostport to Asterisk, it
doesn't run.
Thanks for your support
Adrien
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers