search for: _0043

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

...!! Best regards and greetings from Austria! johnny P.S.: Some detail infos: I'm using Asterisk 1.2.9.1. I installed app_amd from http://www.freedomphones.net/files/app_amd2.c My dialplan context for generating the outgoing calls looks like this (I use AEL): context 1000_amd_tests { _0043[1-9]. => { // be verbose.. :) Verbose(0,InfoInfo ${CONTEXT} - ${EXTEN} called on ${CHANNEL}.); // set our callerId to the correct id Set(CALLERID(number)=xxxxxxxxxx); // absolute call timeout Set(TIMEOUT(absolute)=3600; // debug Verbose(0,DebugDebug Will execute Dial on ${CHANNEL}.);...