[asterisk-users] How do I do this?
I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a telemarketer hold extension which, if I transfer a caller into it, loops around playing music please wait messages, until they give up hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if someone is currently held in that loop. When they hang up, the devstate is re-set the lamp goes out. On serverC, I have a similar devstate, and a couple of extensions - one to turn the lamp on one to turn it off. What happens is this: 1) A call arrives @ Asterisk, and calls a phone on serverA, and a phone on serverC. 2) I answer on serverC, determine it's a telemarketer, and transfer to the telemarketer hold extension on serverA 3) The call enters the loop, and the devstate is set on serverA. As it enters the loop, it calls the turn on extension on serverC, which sets the serverC devstate, and hangs up with an all extensions are busy response. 4) The call, then, stays parked on serverA until the caller hangs up. 5) The h extension on serverA detects the hangup, and re-sets the serverA devstate. 6) Simultaneously, it calls the turn off extension on serverC, which re-sets the devstate returns a all extensions are busy response. 7) serverA then hangs up the call 'officially' by calling Hangup() Unfortunately: Step 6 doesn't do anything on serverC... you can see it being executed on serverA, but the call never arrives at serverC. I'm guessing this is because the caller has already hung up; so, in effect, there's no call to transfer... My question, then, is how to get Asterisk to generate a new call, to tell serverC to switch off it's lamp? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I do this?
- Original Message - From: Ade Vickers [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, December 13, 2007 7:49 AM Subject: [asterisk-users] How do I do this? I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a telemarketer hold extension which, if I transfer a caller into it, loops around playing music please wait messages, until they give up hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if someone is currently held in that loop. When they hang up, the devstate is re-set the lamp goes out. On serverC, I have a similar devstate, and a couple of extensions - one to turn the lamp on one to turn it off. What happens is this: 1) A call arrives @ Asterisk, and calls a phone on serverA, and a phone on serverC. 2) I answer on serverC, determine it's a telemarketer, and transfer to the telemarketer hold extension on serverA 3) The call enters the loop, and the devstate is set on serverA. As it enters the loop, it calls the turn on extension on serverC, which sets the serverC devstate, and hangs up with an all extensions are busy response. 4) The call, then, stays parked on serverA until the caller hangs up. 5) The h extension on serverA detects the hangup, and re-sets the serverA devstate. 6) Simultaneously, it calls the turn off extension on serverC, which re-sets the devstate returns a all extensions are busy response. 7) serverA then hangs up the call 'officially' by calling Hangup() Unfortunately: Step 6 doesn't do anything on serverC... you can see it being executed on serverA, but the call never arrives at serverC. I'm guessing this is because the caller has already hung up; so, in effect, there's no call to transfer... My question, then, is how to get Asterisk to generate a new call, to tell serverC to switch off it's lamp? Cheers, Ade. Use the h exten? Would you mind sharing more details about your setup such as the dialplan or/or apps you are using? I guess you really hate telemarketers ;-) Thanks, Steve Totaro ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I do this?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Totaro
Sent: 13 December 2007 14:35
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How do I do this?
- Original Message -
From: Ade Vickers [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, December 13, 2007 7:49 AM
Subject: [asterisk-users] How do I do this?
I have 2 asterisk servers - serverA and serverC - connected via IAX2.
On serverA, I have a telemarketer hold extension which,
if I transfer a
caller into it, loops around playing music please wait
messages, until
they give up hang up the phone.
Also on serverA, I have a custom devstate, which lights a
lamp on a phone
connected to serverA, which tells me if someone is
currently held in that
loop. When they hang up, the devstate is re-set the lamp goes out.
On serverC, I have a similar devstate, and a couple of
extensions - one to
turn the lamp on one to turn it off.
What happens is this:
1) A call arrives @ Asterisk, and calls a phone on serverA,
and a phone on
serverC.
2) I answer on serverC, determine it's a telemarketer, and
transfer to the
telemarketer hold extension on serverA
3) The call enters the loop, and the devstate is set on
serverA. As it
enters the loop, it calls the turn on extension on
serverC, which sets
the
serverC devstate, and hangs up with an all extensions are
busy response.
4) The call, then, stays parked on serverA until the caller
hangs up.
5) The h extension on serverA detects the hangup, and re-sets the
serverA
devstate.
6) Simultaneously, it calls the turn off extension on
serverC, which
re-sets the devstate returns a all extensions are busy response.
7) serverA then hangs up the call 'officially' by calling Hangup()
Unfortunately: Step 6 doesn't do anything on serverC... you
can see it
being
executed on serverA, but the call never arrives at serverC.
I'm guessing this is because the caller has already hung up; so, in
effect,
there's no call to transfer...
My question, then, is how to get Asterisk to generate a
new call, to
tell
serverC to switch off it's lamp?
Use the h exten? Would you mind sharing more details about
your setup such
as the dialplan or/or apps you are using? I guess you really hate
telemarketers ;-)
Hi Steve,
It's not just telemarketers; I find it's a useful dumping ground for any
caller I don't particularly want to speak to ;)
OK: There are 2 servers involved:
serverA
- Located in the UK, has a connection to a POTS line via an AX100P card.
- Handles any 5xxx extension locally, plus a couple of others
- Talks to serverC via IAX2 channel
- Running Asterisk v1.4.5 + custom devstate patch
serverC
- Located in Spain, has only an internet connection
- Handles any 62xx extension locally, plus the special teledeath_on and
teledeath_off extensions
- Talks to serverA via IAX2 channel
- Running Asterisk v1.4.11 + custom devstate patch
So; in serverA, the following bits of the dialplan are relevant:
[default]
exten = ,hint,custom:telepark
;
-
; When an internal phone dials, this section defines what happens to the
calls
;
-
[internal]
;other destinations cut from here
;Death to telemarketers
exten = ,1,Goto,teledeath|s|1
[teledeath]
exten = s,1,Answer()
exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE)
exten = s,3,Dial(IAX2/serverC/teledeath_on)
exten = s,4,WaitMusicOnHold(15)
exten = s,5,Wait(1)
exten = s,6,Playback(pls-hold-while-try)
exten = s,7,Wait(0.25)
exten = s,8,Goto,4
exten = h,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE)
exten = h,n,Dial(IAX2/serverC/teledeath_off)
exten = h,n,Hangup()
; If anything goes wrong, quit the loop
exten = i,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE)
exten = i,n,Dial(IAX2/serverC/teledeath_off)
exten = i,n,Hangup()
Thus; when I transfer the call to ; it jumps into teledeath|s, which
sets the devstate locally; dials the special extension
IAX2/serverC/teledeath_on to set the serverC busy lamp; then loops around
music/announcements.
When the caller hangs up, teledeath|h is executed; turning off the local
lamp calling IAX2/serverC/teledeath_off - which SHOULD turn off the
serverC lamp, but doesn't - because the call never arrives on serverC...
Here is serverC's extensions.conf file (again, non-pertinent bits removed):
[default]
exten = ,hint,custom:telepark
[internal]
include = outbound
include = default
;Internal phones (local (62xx) remote (everything else)
exten = _[57]XXX,1,Goto,external_extensions|6000${EXTEN}|1
;Death to telemarketers status marker
exten = teledeath_on,1,Set(DEVSTATE(Custom:telepark
Re: [asterisk-users] How do I do this?
Steve Totaro wrote: snippage I suppose you could create a new context on server C, include it in your internal context, and create an h exten on that box to handle it locally. I am unsure why what you have does not work but I assume the unable to transfer is a hint. Except that, once I've transferred the call hung up the serverC end; the call should be entirely handled by serverA; the only further contact the two severs should have in relation to that call is serverA telling serverC to reset the devstate. As you say, the unable to transfer sounds like a clue I wonder if it's due to codecs? Cheers, Ade. No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.503 / Virus Database: 269.17.1/1182 - Release Date: 12/12/2007 11:29 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do I do this?
Ade Vickers wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: 13 December 2007 14:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How do I do this? - Original Message - From: Ade Vickers [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, December 13, 2007 7:49 AM Subject: [asterisk-users] How do I do this? I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a telemarketer hold extension which, if I transfer a caller into it, loops around playing music please wait messages, until they give up hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if someone is currently held in that loop. When they hang up, the devstate is re-set the lamp goes out. On serverC, I have a similar devstate, and a couple of extensions - one to turn the lamp on one to turn it off. What happens is this: 1) A call arrives @ Asterisk, and calls a phone on serverA, and a phone on serverC. 2) I answer on serverC, determine it's a telemarketer, and transfer to the telemarketer hold extension on serverA 3) The call enters the loop, and the devstate is set on serverA. As it enters the loop, it calls the turn on extension on serverC, which sets the serverC devstate, and hangs up with an all extensions are busy response. 4) The call, then, stays parked on serverA until the caller hangs up. 5) The h extension on serverA detects the hangup, and re-sets the serverA devstate. 6) Simultaneously, it calls the turn off extension on serverC, which re-sets the devstate returns a all extensions are busy response. 7) serverA then hangs up the call 'officially' by calling Hangup() Unfortunately: Step 6 doesn't do anything on serverC... you can see it being executed on serverA, but the call never arrives at serverC. I'm guessing this is because the caller has already hung up; so, in effect, there's no call to transfer... My question, then, is how to get Asterisk to generate a new call, to tell serverC to switch off it's lamp? Use the h exten? Would you mind sharing more details about your setup such as the dialplan or/or apps you are using? I guess you really hate telemarketers ;-) Hi Steve, It's not just telemarketers; I find it's a useful dumping ground for any caller I don't particularly want to speak to ;) OK: There are 2 servers involved: serverA - Located in the UK, has a connection to a POTS line via an AX100P card. - Handles any 5xxx extension locally, plus a couple of others - Talks to serverC via IAX2 channel - Running Asterisk v1.4.5 + custom devstate patch serverC - Located in Spain, has only an internet connection - Handles any 62xx extension locally, plus the special teledeath_on and teledeath_off extensions - Talks to serverA via IAX2 channel - Running Asterisk v1.4.11 + custom devstate patch So; in serverA, the following bits of the dialplan are relevant: [default] exten = ,hint,custom:telepark ; - ; When an internal phone dials, this section defines what happens to the calls ; - [internal] ;other destinations cut from here ;Death to telemarketers exten = ,1,Goto,teledeath|s|1 [teledeath] exten = s,1,Answer() exten = s,2,Set(DEVSTATE(Custom:telepark)=INUSE) exten = s,3,Dial(IAX2/serverC/teledeath_on) exten = s,4,WaitMusicOnHold(15) exten = s,5,Wait(1) exten = s,6,Playback(pls-hold-while-try) exten = s,7,Wait(0.25) exten = s,8,Goto,4 exten = h,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE) exten = h,n,Dial(IAX2/serverC/teledeath_off) exten = h,n,Hangup() ; If anything goes wrong, quit the loop exten = i,1,Set(DEVSTATE(Custom:telepark)=NOT_INUSE) exten = i,n,Dial(IAX2/serverC/teledeath_off) exten = i,n,Hangup() Thus; when I transfer the call to ; it jumps into teledeath|s, which sets the devstate locally; dials the special extension IAX2/serverC/teledeath_on to set the serverC busy lamp; then loops around music/announcements. When the caller hangs up, teledeath|h is executed; turning off the local lamp calling IAX2/serverC/teledeath_off - which SHOULD turn off the serverC lamp, but doesn't - because the call never arrives on serverC... Here is serverC's extensions.conf file (again, non-pertinent bits removed): [default] exten = ,hint,custom:telepark [internal
Re: [asterisk-users] How do I do this?
Ade Vickers wrote: I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a telemarketer hold extension which, if I transfer a caller into it, loops around playing music please wait messages, until they give up hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if someone is currently held in that loop. When they hang up, the devstate is re-set the lamp goes out. On serverC, I have a similar devstate, and a couple of extensions - one to turn the lamp on one to turn it off. I know this doesn't really answer your question, but I've achieved excellent inter-asterisk communication using PHP scripts triggered by System(wget http://remote-server/teledeath_off.php;) style stuff, if it helps :) Not sure how that would work with devstate. Maybe the php file would connect to asterisk via the manager interface and dial the teledeath_off extension through the Local channel. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE : [asterisk-users] How do I do this in Asterisk?
Hi Francoies,
Many thanks for your reply will give it a try.
Best regards and thanks,
Christian
On Tue, 2007-05-01 at 20:09 +0200, [EMAIL PROTECTED] wrote:
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten = s,n(menurestart),Background(your_announce)
exten = s,n,WaitExten(5)
exten = 1,1,GoTo(your_menu_context,1,1)
exten = 2,1,GoTo(your_menu_context,2,1)
exten = 3,1,GoTo(your_menu_context,3,1)
exten = t,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = i,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = h,1,NoOp(the caller has hung up)
I hope that can help and to have not introduced mistakes ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de
Christian
Envoyé : mardi 1 mai 2007 18:18
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] How do I do this in Asterisk?
Hi all,
I have created a menu from which the caller can select several
options such as being transfered to our phones and my mobile
phone, meetme, etc. If the caller press an invalid option i
have set it to play a message like invalid choice please try
again. If the caller make three invalid choices i want the
call to be disconnected. what is the best way of doing that?
And finally i have set up an extention to which it is possible
to record a message but i then want to be able to specify what
number the message should be plaied for after recording is
finished. Many thanks for all your help,
Christian
___
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___
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[asterisk-users] How do I do this in Asterisk?
Hi all, I have created a menu from which the caller can select several options such as being transfered to our phones and my mobile phone, meetme, etc. If the caller press an invalid option i have set it to play a message like invalid choice please try again. If the caller make three invalid choices i want the call to be disconnected. what is the best way of doing that? And finally i have set up an extention to which it is possible to record a message but i then want to be able to specify what number the message should be plaied for after recording is finished. Many thanks for all your help, Christian___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [asterisk-users] How do I do this in Asterisk?
Hi Christian,
Increase a variable in the menu loop, or exactly in the t and i
extensions like this :
exten = s,1,Wait(3)
exten = s,n,Answer()
exten = s,n,Set(LoopStep=1)
exten = s,n,Set(TIMEOUT(digit)=3)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Wait(1)
exten = s,n(menurestart),Background(your_announce)
exten = s,n,WaitExten(5)
exten = 1,1,GoTo(your_menu_context,1,1)
exten = 2,1,GoTo(your_menu_context,2,1)
exten = 3,1,GoTo(your_menu_context,3,1)
exten = t,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = i,1,Playback(im-sorry)
exten = t,n,Set(LoopStep=$[${LoopStep} + 1])
exten = t,n,GoToIf($[${LoopStep} 3]?disconnect)
exten = t,n,GoTo(s,menurestart)
exten = t,n(disconnect),Hangup()
exten = h,1,NoOp(the caller has hung up)
I hope that can help and to have not introduced mistakes ;-)
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Christian
Envoyé : mardi 1 mai 2007 18:18
À : asterisk-users@lists.digium.com
Objet : [asterisk-users] How do I do this in Asterisk?
Hi all,
I have created a menu from which the caller can select several options such
as being transfered to our phones and my mobile phone, meetme, etc. If the
caller press an invalid option i have set it to play a message like invalid
choice please try again. If the caller make three invalid choices i want the
call to be disconnected. what is the best way of doing that?
And finally i have set up an extention to which it is possible to record a
message but i then want to be able to specify what number the message should
be plaied for after recording is finished. Many thanks for all your help,
Christian
___
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[Asterisk-Users] How do I do this ?
I wish to initate calls from a web interface, by clicking on a link and then connecting to the automatic outgoing call by picking up an analogue phone. I've got one fxs and one fxo and I wish to automate the call using a call file (which I can do now). How can I pick up a handset and connect to this call I've made when it's ringing? Can someone point me as to how I may be able to do this. ___ 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
Re: [Asterisk-Users] How do I do this ?
FOP http://www.asternic.org/ On Tue, 22 Feb 2005 22:40:32 +1100, PHP Mechanic [EMAIL PROTECTED] wrote: I wish to initate calls from a web interface, by clicking on a link and then connecting to the automatic outgoing call by picking up an analogue phone. I've got one fxs and one fxo and I wish to automate the call using a call file (which I can do now). How can I pick up a handset and connect to this call I've made when it's ringing? Can someone point me as to how I may be able to do this. ___ 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
RE: [Asterisk-Users] How do I do this ?
Me agree too. PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, 23 February 2005 12:37 AM To: PHP Mechanic; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How do I do this ? FOP http://www.asternic.org/ On Tue, 22 Feb 2005 22:40:32 +1100, PHP Mechanic [EMAIL PROTECTED] wrote: I wish to initate calls from a web interface, by clicking on a link and then connecting to the automatic outgoing call by picking up an analogue phone. I've got one fxs and one fxo and I wish to automate the call using a call file (which I can do now). How can I pick up a handset and connect to this call I've made when it's ringing? Can someone point me as to how I may be able to do this. ___ 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 CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ 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
RE: [Asterisk-Users] how do I do s extensions with PRI
Put the _X below the first 4 extensions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 5:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how do I do s extensions with PRI I would like to know how to define the s extension when I have an incoming PRI line? Currently I have 5 incoming DID numbers. Four of these DID numbers I have going to specific extensions, the fifth number which is the main number I wish to go to a background sound where callers can hear message, get directory, dial extension, whatever. I see that the way to normally do this would be to define s extensions and then step up the priorities for each action I wished to be taken. However, with the PRI line it seems that I can't use the s extension. I can use exten = _X. but this screws up the other four DID numbers which I have going to specific extensions. Is there a way with a PRI that I can define an s extension or something like it to save from having to type an entire 10 digit string multiple places? Thanks for any suggestions. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how do I do s extensions with PRI
The main number extension (_X) is in a different context than the other 4 extensions plus what do I do with the timeout arguement and other things such as that? Just point them to _X.? AJ On Wed, 23 Jul 2003, Todd Lieberman wrote: Put the _X below the first 4 extensions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, July 23, 2003 5:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] how do I do s extensions with PRI I would like to know how to define the s extension when I have an incoming PRI line? Currently I have 5 incoming DID numbers. Four of these DID numbers I have going to specific extensions, the fifth number which is the main number I wish to go to a background sound where callers can hear message, get directory, dial extension, whatever. I see that the way to normally do this would be to define s extensions and then step up the priorities for each action I wished to be taken. However, with the PRI line it seems that I can't use the s extension. I can use exten = _X. but this screws up the other four DID numbers which I have going to specific extensions. Is there a way with a PRI that I can define an s extension or something like it to save from having to type an entire 10 digit string multiple places? Thanks for any suggestions. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how do I do s extensions with PRI
The way I do this is to create a separate context that has a
match-all statement, and then jump to a supplemental context that
does the real matching. Just use ${EXTEN} in the Goto jump. You can
also just put the _X. in front of the other extensions in the same
context, but I like to keep things a little bit apart to prevent
ordering confusion.
[generic1]
exten = _X.,1,SetVar(FOO=BAR)
exten = _X.,2,BogoApplication(some-variables)
exten = _X.,3,Goto(generic2,${EXTEN},1)
[generic2]
exten = _345.,1,Dial(SIP/345)
.
.
.
(etc. - all of your real matches in here)
JT
I would like to know how to define the s extension when I have an incoming
PRI line? Currently I have 5 incoming DID numbers. Four of these DID
numbers I have going to specific extensions, the fifth number which is the
main number I wish to go to a background sound where callers can hear
message, get directory, dial extension, whatever. I see that the way to
normally do this would be to define s extensions and then step up the
priorities for each action I wished to be taken. However, with the PRI
line it seems that I can't use the s extension. I can use exten = _X.
but this screws up the other four DID numbers which I have going to
specific extensions. Is there a way with a PRI that I can define an s
extension or something like it to save from having to type an entire 10
digit string multiple places?
Thanks for any suggestions.
AJ
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Re: [Asterisk-Users] how do I do s extensions with PRI
A PRI Does not have a s extension as all calls will have a DID assigned to the call. At best, on your 5th DID place a Goto to the s|1 and be done with it. On Wed, 2003-07-23 at 16:14, [EMAIL PROTECTED] wrote: I would like to know how to define the s extension when I have an incoming PRI line? Currently I have 5 incoming DID numbers. Four of these DID numbers I have going to specific extensions, the fifth number which is the main number I wish to go to a background sound where callers can hear message, get directory, dial extension, whatever. I see that the way to normally do this would be to define s extensions and then step up the priorities for each action I wished to be taken. However, with the PRI line it seems that I can't use the s extension. I can use exten = _X. but this screws up the other four DID numbers which I have going to specific extensions. Is there a way with a PRI that I can define an s extension or something like it to save from having to type an entire 10 digit string multiple places? Thanks for any suggestions. AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
