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

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(a)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 : Otherwise plan you Ofcourse you check the SER to an sip:enum@domain.tld extension. Asterisk, it