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@gwiaxt01.voicepulse.com${EXTEN})
;VoicePulse Connect 2
exten =>
_1NXXNXXXXXX,2,Dial(IAX2/UserName:Password@gwiaxt02.voicepulse.com${EXTEN})
;Nufone
exten => _1NXXNXXXXXX,3,Dial,IAX2/UserName@NuFone/${EXTEN}
exten => t,1,Hangup
****************************************************************************
**********
Thanks,
Todd Routhier
Lightwave Technologies, LLC.
--
Start Your Dialup Internet Service!
http://www.YourOwnISP.com
Lightwave Technologies, LLC.
http://www.LightWaveTech.com
at I have in my extensions.conf file> below minus my auth info :) > > What happens now is this, if I pickup the sip phone at ext. 100 and > dial extension 101 the phone at 101 rings but when 101 answers we > can't talk between the phones it's silence.Check: Canreinvite=$value Codecs are the same on both phones
>Message: 16 >Date: Tue, 19 Oct 2004 13:13:52 -0500From: "Your Own ISP .com" <mylist@lightwavetech.com>>Subject: [Asterisk-Users] How to ring internal extension? >To: <asterisk-users@lists.digium.com> >Message-ID: <20041019181344.1A1BD2FC8CF@lists.digium.com> >Content-Type: text/plain; charset="US-ASCII" > >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.This is a stab in the dark, but can it have anything to do with your 'm' option by the dial? Do you have any musiconhold working on 100? Also, I'm assuming in sip.conf, you've specified context as outgoing (I don't think it would ring otherwise).>[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
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: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Your Own ISP
.com
Sent: Tuesday, October 19, 2004 1:14 PM
To: asterisk-users@lists.digium.com
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:Password@gwiaxt01.voicepulse.com${EXTEN})
;VoicePulse Connect 2
exten =>
_1NXXNXXXXXX,2,Dial(IAX2/UserName:Password@gwiaxt02.voicepulse.com${EXTEN})
;Nufone
exten => _1NXXNXXXXXX,3,Dial,IAX2/UserName@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
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>Message: 9 >Date: Tue, 19 Oct 2004 14:03:03 -0500From: "Your Own ISP .com" <mylist@lightwavetech.com>>Subject: RE: [Asterisk-Users] How to ring internal extension? >To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> >Message-ID: <20041019190257.374901C0DC@lists.digium.com> >Content-Type: text/plain; charset="US-ASCII" > >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. You can write include => outgoing in your incoming, and then anything able to be done by outgoing should be able to be done in incoming...see where that gets you.