[Asterisk-Users] How to ring internal extension?

[Your Own ISP .com]

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:Password at gwiaxt01.voicepulse.com${EXTEN})

;VoicePulse Connect 2
exten =>
_1NXXNXXXXXX,2,Dial(IAX2/UserName:Password at gwiaxt02.voicepulse.com${EXTEN})


;Nufone
exten => _1NXXNXXXXXX,3,Dial,IAX2/UserName at NuFone/${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