On 7/17/07, inge <inge@legos.fr> wrote:

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@dimain.tld") {
>
>     rewritehost("asteriskip") ;  //something like this. check the
> syntax.
>    t_relay();
>     break;
>  };
>
> Hope this helps,
>
> On 7/17/07, inge <inge@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
>
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