Also, do I need to duplicate my extension info under each section (outgoing & incoming) or is there a better way?
Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Your Own ISP .com Sent: Tuesday, October 19, 2004 1:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to ring internal extension? So far I have managed to get a working system up and running for calling from a Sip phone and out to a Termination Provider to the PSTN as well as calling a termination provider DID from the PSTN and having the call go through to my Sip phone. What I want to do now is simply pick up 1 sip phone at say extension 100 and dial another sip phone at extension 101 and connect the calls without the termination provider in the middle. I am unclear on how to do this and I am not sure where to look for this info. I have pasted my basic setup that I have in my extensions.conf file below minus my auth info :) What happens now is this, if I pickup the sip phone at ext. 100 and dial extension 101 the phone at 101 rings but when 101 answers we can't talk between the phones it's silence. As I watch the Asterisk console everything seems to look fine, it mentions the dialing then the setting up of a native bridge etc. Any idea what I have done wrong here? I suspect there is a MUCH better way to go about this that I am totally missing, this is just what I hacked together by trial and error. ************************************************* [FromVoicePulse] ; <-- Should match the context you have ; under [voicepulse-in-01] in iax.conf exten => _NXXNXXXXXX,1,Answer exten => _NXXNXXXXXX,2,Background(ext-or-zero) exten => _NXXNXXXXXX,3,DigitTimeout,3 exten => _NXXNXXXXXX,4,ResponseTimeout,30 ;Operator exten => 0,1,Answer exten => 0,2,Background(tt-weasels) exten => 0,3,DigitTimeout,3 exten => 0,4,ResponseTimeout,20 ; 100 - Todd's Voicemail exten => 100,1,Dial(SIP/100,30,m) exten => 100,2,Goto,t|1 ; 101 - Lewis' Voicemail exten => 101,1,Dial(SIP/101,30,m) exten => 101,2,Goto,t|1 ;exten => t,1,Playback,vm/generic/goodbye exten => t,1,Hangup ************************************************* >> Then I have something like this in the extensions.conf for outgoing: >> ************************************************* [outgoing] ; 100 - Todd's Voicemail exten => 100,1,Dial(SIP/100,15,m) ;exten => 100,1,Playback,vm/100/unavail ;exten => 100,2,Voicemail,1 exten => 100,2,Goto,t|1 ; 101 - Lewis' Voicemail exten => 101,1,Dial(SIP/101,15,m) ;exten => 101,1,Playback,vm/101/unavail ;exten => 101,2,Voicemail,1 exten => 101,2,Goto,t|1 ;VoicePulse Connect 1 exten => _1NXXNXXXXXX,1,Dial(IAX2/UserName:[EMAIL PROTECTED]) ;VoicePulse Connect 2 exten => _1NXXNXXXXXX,2,Dial(IAX2/UserName:[EMAIL PROTECTED]) ;Nufone exten => _1NXXNXXXXXX,3,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN} exten => t,1,Hangup **************************************************************************** ********** Thanks, Todd Routhier Lightwave Technologies, LLC. -- Start Your Dialup Internet Service! http://www.YourOwnISP.com Lightwave Technologies, LLC. http://www.LightWaveTech.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users