Re: Problems Solved, two left

Doug Lytle <[email protected]> Wed, 24 May 2023 09:16:41 -0400
Newsgroups gmane.comp.telephony.pbx.asterisk.user
Message-ID <[email protected]>
On 5/24/23 08:03, Steve Matzura wrote:
>
> ***  extensions.conf  ***
>
>
> [general]
>
> [globals]
>
> ; Make sure to include inbound prior to outbound because the 
> _NXXNXXXXXX handler will match the incoming call and create a loop
> include => voipms-inbound
> include => voipms-outbound
>
> [voipms-outbound]
> exten => _1NXXNXXXXXX,1,Dial(PJSIP/${EXTEN}@voipms)
> exten => _1NXXNXXXXXX,n,Hangup()
> exten => _NXXNXXXXXX,1,Dial(PJSIP/1${EXTEN}@voipms)
> exten => _NXXNXXXXXX,n,Hangup()
> exten => _011.,1,Dial(PJSIP/${EXTEN}@voipms)
> exten => _011.,n,Hangup()
> exten => _00.,1,Dial(PJSIP/${EXTEN}@voipms)
> exten => _00.,n,Hangup()
>
> ; inbound context example for your DID numbers, do not add the number 
> 1 in front
>
> [voipms-inbound]
> exten => {redacted},1,Goto(hello,200,1) ; My  DID
>
> [phones]
> exten => 101,1,Dial(PJSIP/yealink)
>
> [hello]
> exten => 200,1,Answer()
>     same => n,Playback(hello-world)
>     same => n,Hangup()
>


Your inbound is being sent to s (start extension) instead of your DID, 
so it's not matching.  So, you'll need to find out where in your 
dialplan it's being mapped to s.

Did you know that voip.ms supports IAX2 natively?  Working much better, 
in my opinion, that SIP.

Doug

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users