[Asterisk-Users] gs on phone ?

Chris HARIGA contact at techselesta.com
Tue Mar 2 20:36:45 MST 2004


The right conf must be like this:

 exten => 2015,1,Dial(SIP/2015 at 2015,20,T,t)
 exten => 2015,2,Voicemail(u${EXTEN})
 exten => 2015,102,Voicemail(b${EXTEN})
 exten => 2015,103,Hangupv

Chris HARIGA

Techselesta Inc.
http://www.techselesta.com/


----- Original Message ----- 
From: "Chris Clifton" <chris at netlabz.com>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, March 02, 2004 10:28 PM
Subject: [Asterisk-Users] gs on phone ?


>
> I have a GS101 connected to * with sip and g729.
>
> When an incoming call comes in from outside (via pstn for example), and no
> one picks up the GS, * reports that 'the user is on the phone'. If no one
> answers, I'd expect it to report 'unavailable'.
>
> Maybe I'm not understanding the call flow ... (should it be u$ at '2',
then
> b$ at '102' ?) My current config for call flow seems to match others I've
> seen on the wiki, etc.
>
> my extensions.conf for the grandstream at x2015 -
>
> [incoming]
> exten => 2015,1,Dial(SIP/2015 at 2015,20,T,t)
> exten => 2015,2,Voicemail(b${EXTEN})
> exten => 2015,3,Hangup
> exten => 2015,102,Voicemail(u${EXTEN})
> exten => 2015,103,Hangup
>
> Thanks,
> Chris Clifton
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>





More information about the asterisk-users mailing list