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@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
The right conf must be like this:
exten => 2015,1,Dial(SIP/2015@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@netlabz.com>
To: <asterisk-users@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@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@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
In your extensions.conf, the "b" and "u" are reversed. Use
u${EXTEN} for
priority 2 and b${EXTEN} for priority 102.
-Ron
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Chris Clifton
Sent: Tuesday, March 02, 2004 10:29 PM
To: asterisk-users@lists.digium.com
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@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
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