Hi All, 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 phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James
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=dynamic callerid=9500 disallow=all allow=ulaw allow=alaw dtmfmode=info mailbox=9500 callgroup=1 pickupgroup=1 cancallforward=yes Craig ----- Original Message ----- From: "James Freire" <JFreire@Comtech21.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, August 21, 2004 12:09 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 for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
----- Original Message ----- From: "James Freire" <JFreire@Comtech21.com> To: <asterisk-users@lists.digium.com> 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 PBXfeatures to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions.> > Thanks, > > JamesSomeone correct me if I'm wrong but I believe you'll need the dialplan for this one... What I envision is doing something like this... [verticalservice] exten => *78,1,DbGet(${dnd}=features/dnd) exten => *78,2,DbPut(features/dnd=1) exten => *78,3,Playback(pbx-dndenabled) exten => *78,4,Hangup() exten => *78,102,GotoIf($[${dnd} = '0')]?103:104) exteh => *78,103,DbPut(features/dnd=1) exten => *78,104,Playback(pbx-dndenabled) exten => *78,105,Hangup() exten => *79 ... etc... Then in your extension calling macro, you're going to want to check against the DB like this... [macro-insidedial] exten => s,1,DbGet(${dnd}=features/dnd) exten => s,2,DbGet(${fw}=features/fw) exten => s,3,Dial(${ARG1},25,tT) exten => s,4,VoiceMail(u${ARG1}) exten => s,5,Hangup() exten => s,102,GotoIf($[${dnd} = '1']?200:2) exten => s,103,GotoIf($[${fw} = '1']?300:3) exten => s,104,VoiceMail(b${ARG1}) exten => s,200,VoiceMail(b${ARG1}) exten => s,201,Hangup() exten => s,300,Dial(SIP/${fw}c@sipprovider-out,60) exten => s,301,Congestion() be sure to include [verticalservice] in your inside-office context...
Hi Craig,
Thank you very much for the helpful information. I did enable that setting and
it seems to have worked but not all the way. I do a *72 for an unconditional
call forward + the number to forward to. Then when I dial the grandstream that
has it enabled, asterisk just reponds that the extension is busy, the BT does
not foward the call. I also get the following on the CLI
-- Executing Dial("Zap/8-1", "SIP/2000|20") in new stack
-- Called 2000
-- Got SIP response 302 "Moved Temporarily" back from 64.201.13.50
-- SIP/2000-42e8 is busy
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Craig Guy
Sent: Friday, August 20, 2004 12:36 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
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=dynamic
callerid=9500
disallow=all
allow=ulaw
allow=alaw
dtmfmode=info
mailbox=9500
callgroup=1
pickupgroup=1
cancallforward=yes
Craig
----- Original Message -----
From: "James Freire" <JFreire@Comtech21.com>
To: <asterisk-users@lists.digium.com>
Sent: Saturday, August 21, 2004 12:09 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 for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.
Thanks,
James
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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 doing something like this... > > [verticalservice] > > exten => *78,1,DbGet(${dnd}=features/dnd) > exten => *78,2,DbPut(features/dnd=1) > exten => *78,3,Playback(pbx-dndenabled) > exten => *78,4,Hangup() > exten => *78,102,GotoIf($[${dnd} = '0')]?103:104) > exteh => *78,103,DbPut(features/dnd=1) > exten => *78,104,Playback(pbx-dndenabled) > exten => *78,105,Hangup() > > exten => *79 ... etc...Wouldn't you need to track each extension? something like: exten => *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM}) exten => *78,2,DbPut(dnd/${CALLERIDNUM}=1) exten => *78,3,Playback(pbx-dndenabled) exten => *78,4,Hangup() etc.? The wiki has an exmple for call forwarding: http://www.voip-info.org/wiki-Asterisk+call+forwarding _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
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" <JFreire@Comtech21.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 20, 2004 6:09 PM 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 for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James _______________________________________________ Asterisk-Users mailing list Asterisk-Users@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
http://www.voip-info.org/wiki-asterisk+pbx+functions http://www.voip-info.org/wiki-asterisk+vertical+service+activation+codes ----- Original Message ----- From: "Robert Rozman" <rozman@fri.uni-lj.si> To: <asterisk-users@lists.digium.com> Sent: Friday, August 20, 2004 3:02 PM Subject: Re: [Asterisk-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 ----- > From: "James Freire" <JFreire@Comtech21.com> > To: <asterisk-users@lists.digium.com> > Sent: Friday, August 20, 2004 6:09 PM > 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 for DND, call foward, etc. These functions do work when I > use my POTS phones hooked up to my Zap cards. But I cannot get the PBX > functions (ie *78, *79) to work using my SIP phones. Is there a featurethat> has to be enabled to do this? I know these functions are available within > the GS phone but all of them seem to just show the phone as being busy,even> though, say, call foward is supposed to foward. It just makes the phone > busy. I figure it would be easier just to have asterisk handling all those > PBX functions. > > Thanks, > > James > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@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