[Serusers] Transfering incoming call on SER to Asterisk

[inge]

Hi,

I don't have Asterisk configuration. Sorry, it's a customer which
configure it.

But for SER configuration I do something like that :

rewritehost("IPASTERISK");
t_relay();
break;

We succeeded in doing a trunk for incoming and outgoing calls.

I hope that can help you.

Sincerely,

Adrien .L


Le lundi 23 juillet 2007 à 09:55 -0700, Jai Rangi a écrit :
> 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
>         >
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>