Craig Guy wrote:
cancallforward=yes
There is no such function in distributed chan_sip.c,
ergo there can't be such a configuration parameter.
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi James,
This
terisk-Users] Asterisk PBX Functions via SIP phone
> Hi,
>
> sorry for interruption, but are there any guides for all possible Asterisk
> PBX functions that are available with no particular dialplan handling ?
>
> Thanks,
>
> Robert.
>
> - Original Message -
Hi,
sorry for interruption, but are there any guides for all possible Asterisk
PBX functions that are available with no particular dialplan handling ?
Thanks,
Robert.
- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 6:0
- Original Message -
From: "Olle E. Johansson" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 1:37 PM
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
> Chris Shaw wrote:
>
> >>I am suprised that one woul
Chris Shaw wrote:
I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?
I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service
> I am suprised that one would have to create a dialplan since its an
already built in function that works with regular POTS phones. Or is it
because of the way DTMF is sent via SIP?
I don't know digium's long range plans, but looking through chan_sip.c NONE
of the vertical service codes are menti
I am suprised that one would have to create a dialplan since its an already built in
function that works with regular POTS phones. Or is it because of the way DTMF is sent
via SIP?
> Someone correct me if I'm wrong but I believe you'll need the dialplan for
> this one...
>
> What I envision is
> Wouldn't you need to track each extension? something like:
> exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
> exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1)
> exten => *78,3,Playback(pbx-dndenabled)
> exten => *78,4,Hangup()
> etc.?
>
Yep! good catch! that's why I asked someone to correct me, I was
On Fri, Aug 20, 2004 at 10:13:16AM -0700, Chris Shaw said:
> - Original Message -
> From: "James Freire" <[EMAIL PROTECTED]>
>
> > I am using a Grandstream BT100 and I have been trying to get the PBX
> features to work for DND, call foward, etc. These functions do work when I
> use my POTS
Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi James,
This is a feature that needs to be enabled on both the pho
- Original Message -
From: "James Freire" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Friday, August 20, 2004 9:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
> Hi All,
>
> I am using a Grandstream BT100 and I have been trying to get the PBX
features to work fo
Hi James,
This is a feature that needs to be enabled on both the phones and on
Asterisk. So after enabling on your BT100 you need to add
'cancallforward=yes' to each extension in sip.conf you would like to add
this feature to as in :-
[9500]
context=internal
type=friend
username=9500
host=dynami
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