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 >