Folks,
Apologies ahead of time if this has already been asked (read the list for
the last month looking
for something similar).
I have been trying to get the Background command to work with no joy yet.
Here is what I am trying to do:
1. Answer the call.
2. Play the message in the background, while waiting on DTMF from user.
3. If I get a "1", then interrupt the message and dial the phone
number listed
at extension "1", otherwise play "invalid extension"
and cycle back.
Here is what I have so far:
* note: phone numbers changed to "generic"
[message]
exten => s,1,Answer()
exten => s,2,Wait(2) ;Pause to let the user end catch up with the connection
exten => s,3,Background(demo-congrats)
exten => s,4,Goto(3)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,3)
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(SIP/4805551212/20)
[incoming]
exten => 8665551212,1,Goto(message,s,1)
What I get is:
1. The number is answered and the demo-congrats file plays.
2. No matter what button I press on the phone, the file continues to play
and recycle when it's done.
Question(s):
1. Is there a proper way to configure this?
2. Am I missing a configuration step somewhere in the one of the conf files.
Thanks in advance,
...Paul
Use Read instead of background
CLI> show application Read
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Paul Penrod
Sent: Friday, September 17, 2004 6:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Background() command
Folks,
Apologies ahead of time if this has already been asked (read the list for
the last month looking
for something similar).
I have been trying to get the Background command to work with no joy yet.
Here is what I am trying to do:
1. Answer the call.
2. Play the message in the background, while waiting on DTMF from user.
3. If I get a "1", then interrupt the message and dial the phone
number
listed
at extension "1", otherwise play "invalid extension"
and cycle back.
Here is what I have so far:
* note: phone numbers changed to "generic"
[message]
exten => s,1,Answer()
exten => s,2,Wait(2) ;Pause to let the user end catch up with the
connection
exten => s,3,Background(demo-congrats)
exten => s,4,Goto(3)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,3)
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(SIP/4805551212/20)
[incoming]
exten => 8665551212,1,Goto(message,s,1)
What I get is:
1. The number is answered and the demo-congrats file plays.
2. No matter what button I press on the phone, the file continues to play
and recycle when it's done.
Question(s):
1. Is there a proper way to configure this?
2. Am I missing a configuration step somewhere in the one of the conf files.
Thanks in advance,
...Paul
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Here's how I handle your situation and it works just fine: [usersmenu] exten => s,1,DigitTimeout,5 ;set digit timeout to 5 seconds exten => s,2,ResponseTimeout,10 ; Set Response Timeoute to 10 sec exten => s,3,Background(usermenuopts) ; Press 1 for lyle, 2 for Adam, 3 for Lois or timeout to next step exten => s,4,Voicemail(u110); Drop into general vm for LCR exten => 1,1,Voicemail(u101) exten => 2,1,Voicemail(u103) exten => 3,1,Voicemail(u102) exten => 4,1,Voicemail(u110) exten => 0,1,Goto(usersmenu,s,1) Allthough I finish differently, hitting 1,2,3,4,0 or any valid extension during the background works just fine. Lyle ----- Original Message ----- From: "Paul Penrod" <P.Penrod@fiber-ride.com> To: <asterisk-users@lists.digium.com> Sent: Friday, September 17, 2004 5:52 PM Subject: [Asterisk-Users] Background() command> Folks, > > Apologies ahead of time if this has already been asked (read the list for > the last month looking > for something similar). > > I have been trying to get the Background command to work with no joy yet. > > Here is what I am trying to do: > > 1. Answer the call. > 2. Play the message in the background, while waiting on DTMF from user. > 3. If I get a "1", then interrupt the message and dial the phone numberlisted> at extension "1", otherwise play "invalid extension" and cycle back. > > Here is what I have so far: > > * note: phone numbers changed to "generic" > > [message] > exten => s,1,Answer() > exten => s,2,Wait(2) ;Pause to let the user end catch up with theconnection> exten => s,3,Background(demo-congrats) > exten => s,4,Goto(3) > > exten => t,1,Playback(vm-goodbye) > exten => t,2,Hangup() > > exten => i,1,Playback(pbx-invalid) > exten => i,2,Goto(s,3) > > exten => 1,1,Playback(transfer) > exten => 1,2,Dial(SIP/4805551212/20) > > [incoming] > > exten => 8665551212,1,Goto(message,s,1) > > What I get is: > > 1. The number is answered and the demo-congrats file plays. > 2. No matter what button I press on the phone, the file continues to play > and recycle when it's done. > > Question(s): > > 1. Is there a proper way to configure this? > 2. Am I missing a configuration step somewhere in the one of the conffiles.> > Thanks in advance, > > ...Paul > > _______________________________________________ > 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 >
On 17 Sep 2004 at 16:52, Paul Penrod wrote:> 1. The number is answered and the demo-congrats file plays. > 2. No matter what button I press on the phone, the file continues to > play and recycle when it's done.What you could do is debug <channel name> in the asterisk console one the call has come in (just use the tab key to find the channel once the call is up and after typing debug - i.e. debug <tabkey>). Then you should be presented with a message whenever a DTMF tone is received. I would say that maybe you are using a compressed codec with inband DTMF signalling maybe? Matt Riddell (New Zealand Digium Distribution/Custom Software) http://www.sineapps.com/contact.php (contact us) http://www.sineapps.com/news.php (Daily Asterisk News)
On Fri, 17 Sep 2004, Paul Penrod wrote:> [message] > exten => s,1,Answer() > exten => s,2,Wait(2) ;Pause to let the user end catch up with the connection > exten => s,3,Background(demo-congrats) > exten => s,4,Goto(3)Remove the goto. Steve
I have a similar dial plan that works just fine. What I did find is that my
sip phones were set to play DTMF tones "in-band" but they needed to be
rfc
(think this setting has to line up with your setting in sip.conf). If they
don't match, you can press keys until the cows come home but Asterisk will
not hear them.
Bill Seddon
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Penrod
Sent: September 17, 2004 11:53 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Background() command
Folks,
Apologies ahead of time if this has already been asked (read the list for
the last month looking
for something similar).
I have been trying to get the Background command to work with no joy yet.
Here is what I am trying to do:
1. Answer the call.
2. Play the message in the background, while waiting on DTMF from user.
3. If I get a "1", then interrupt the message and dial the phone
number
listed
at extension "1", otherwise play "invalid extension"
and cycle back.
Here is what I have so far:
* note: phone numbers changed to "generic"
[message]
exten => s,1,Answer()
exten => s,2,Wait(2) ;Pause to let the user end catch up with the
connection
exten => s,3,Background(demo-congrats)
exten => s,4,Goto(3)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(s,3)
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(SIP/4805551212/20)
[incoming]
exten => 8665551212,1,Goto(message,s,1)
What I get is:
1. The number is answered and the demo-congrats file plays.
2. No matter what button I press on the phone, the file continues to play
and recycle when it's done.
Question(s):
1. Is there a proper way to configure this?
2. Am I missing a configuration step somewhere in the one of the conf files.
Thanks in advance,
...Paul
_______________________________________________
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