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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    Serusers@lists.iptel.org
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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@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
    >
    >         _______________________________________________
    >         Serusers mailing list
    >         Serusers@lists.iptel.org
    >         http://lists.iptel.org/mailman/listinfo/serusers
    >