On Thu, 22 Sep 2005 [EMAIL PROTECTED] wrote: > Hello, > > i wonder why i didn't find a solution for this problem yet, because it > seems very common: > > I have an asterisk server with an AVM (Fritz) ISDN-Card (BRI), and some > SIP-Softphones which i can call from outside by calling the phonenumber of the > Asterisk-Server and then dialing the number of the SIP-Phone. > If I make a call from a SIP-Phone into PSTN, only the MSN of the > asterisk-server > is submitted, without the extension of the SIP-Phone. > I tried to give Asterisk several MSNs in "capi.conf" and to dial in > "extension.conf" like the following: > exten => _0.,1,Dial(CAPI/@<ASTERISK-MSN><SIP-EXTENSION>:${EXTEN}) > This was just for testing, so i used a fix <SIP-EXTENSION>, but just > the <ASTERISK-MSN> was submitted. > > Is there a way to submit the whole number ? Is it generally possible to do > this with a BRI-Card ?
No, not with a PtMP (MSN based) connection. Only the MSNs given by your provider are possible. If you have more MSNs, you can use one MSN per SIP-phone. What you want to do, can be done with a PtP (DID) connection. PtP provides a base number like 1234 and extensions, e.g. 1234-11. Armin _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users