[Serusers] Transfering incoming call on SER to Asterisk

Jai Rangi jprangi at gmail.com
Mon Jul 23 18:55:47 CEST 2007


Can you post your configurations and ngrep logs.
We use asterisk and ser for our calling application and dont have any
issues.


-Jai
www.bingotelecom.com




On 7/17/07, inge <inge at 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 at dimain.tld") {
> >
> >     rewritehost("asteriskip") ;  //something like this. check the
> > syntax.
> >    t_relay();
> >     break;
> >  };
> >
> > Hope this helps,
> >
> > On 7/17/07, inge <inge at 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 at 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 at lists.iptel.org
> >         http://lists.iptel.org/mailman/listinfo/serusers
> >
>
>
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