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salah.elharit200 at gm...
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 Posted: Wed Nov 27, 2013 4:57 pm Post subject: [asterisk-users] issue with speech in IVR |
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|
hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}music1)
exten => s,n,Background(${sounds_path}music2)
exten => s,n,Background(${sounds_path}music3)
exten => s,n,WaitExten(5)
exten => s,n,goto(home,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,WaitExten(5)
exten => i,n,goto(home,s,1)
exten => 1,1,Goto(project,s,1)
[project]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}mymusic)
exten => s,n,WaitExten(5)
exten => s,n,Goto(project,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards |
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emilianovazquez at gma...
Guest
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| Posted: Wed Nov 27, 2013 7:03 pm Post subject: [asterisk-users] issue with speech in IVR |
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Go inside the time machine and come back to 2013!
Use a newer version asterisk and you will get help. There are a lot of changes and a lot of bugs solved.
Best regards.
Emiliano
Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/)
-----Original Message-----
From: Salaheddine Elharit <salah.elharit200@gmail.com>
Sender: asterisk-users-bounces@lists.digium.comDate: Wed, 27 Nov 2013 21:57:35
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: [asterisk-users] issue with speech in IVR
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paul.belanger at polyb...
Guest
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| Posted: Wed Nov 27, 2013 7:30 pm Post subject: [asterisk-users] issue with speech in IVR |
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On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
| Quote: | hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}music1)
exten => s,n,Background(${sounds_path}music2)
exten => s,n,Background(${sounds_path}music3)
exten => s,n,WaitExten(5)
exten => s,n,goto(home,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,WaitExten(5)
exten => i,n,goto(home,s,1)
exten => 1,1,Goto(project,s,1)
|
exten => _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
| Quote: |
[project]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}mymusic)
exten => s,n,WaitExten(5)
exten => s,n,Goto(project,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
|
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
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salah.elharit200 at gm...
Guest
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| Posted: Thu Nov 28, 2013 5:23 am Post subject: [asterisk-users] issue with speech in IVR |
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hello,
i have add the the code below but the issue still the same i can't go to the project during the speech
any other solution
best regards
NB:for the version of asterisk i can't move to another version for the moment
exten => _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
2013/11/28 Paul Belanger <paul.belanger@polybeacon.com (paul.belanger@polybeacon.com)>
| Quote: | On 13-11-27 04:57 PM, Salaheddine Elharit wrote:
| Quote: | hello list
i have an IVR menu in asterisk 1.4
like below
exten => 600,1,Ringing()
exten => 600,n,Wait(2)
exten => 600,n,Goto(home,s,1)
[home]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}music1)
exten => s,n,Background(${sounds_path}music2)
exten => s,n,Background(${sounds_path}music3)
exten => s,n,WaitExten(5)
exten => s,n,goto(home,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,WaitExten(5)
exten => i,n,goto(home,s,1)
exten => 1,1,Goto(project,s,1)
|
exten => _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
| Quote: |
[project]
exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/)
exten => s,n,Background(${sounds_path}mymusic)
exten => s,n,WaitExten(5)
exten => s,n,Goto(project,s,1)
exten => i,1,Playback(${sounds_path}error)
exten => i,n,goto(project,s,1)
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
|
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com (paul.belanger@polybeacon.com) | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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asterisk_list at earth...
Guest
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| Posted: Thu Nov 28, 2013 6:18 am Post subject: [asterisk-users] issue with speech in IVR |
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On Wednesday 27 November 2013, Salaheddine Elharit wrote:
| Quote: | hello list
i have an IVR menu in asterisk 1.4
[stuff deleted]
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
|
This is an actual dialplan application that I wrote. It's a "spike" -- a
proof of concept that is all depth and no breadth. It's known to work in
Asterisk 1.8.
The sound files "ajs_juke01" and "ajs_anykey" you will need to create for
yourself, depending what MP3s you have available (and replace ajs_ with your
own prefix). You can interrupt the announcements or the MP3s by pressing keys
while playing.
;;;;;;;;;;; VERY PRIMITIVE JUKE BOX CONTEXT ;;;;;;;;;;;
[vpjb]
exten => s,1,Background(ajs_juke01)
; "Press 1 for Ocean Colour Scene, 2 for Crowded House"
exten => s,n,WaitExten(1)
exten => s,n,Goto(1)
exten => i,1,Hangup()
exten => 1,1,Background(ajs_anykey)
; "Press any key to stop the music and return to the menu"
exten => 1,n,MP3Player(/songs/09_policemen+pirates.mp3)
exten => 1,n,Goto(vpjb,s,1)
exten => 2,1,Background(ajs_anykey)
; "Press any key to stop the music and return to the menu"
exten => 2,n,MP3Player(/songs/15_distant_sun.mp3)
exten => 2,n,Goto(vpjb,s,1)
exten => _X,1,Hangup()
--
AJS
Answers come *after* questions.
--
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salah.elharit200 at gm...
Guest
|
| Posted: Thu Nov 28, 2013 10:37 am Post subject: [asterisk-users] issue with speech in IVR |
|
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hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context
any other idea please
best regards .
2013/11/28 A J Stiles <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)>
| Quote: | On Wednesday 27 November 2013, Salaheddine Elharit wrote:
| Quote: | hello list
i have an IVR menu in asterisk 1.4
|
| Quote: | [stuff deleted]
my problem when the customor call the number 600 and press 1 in order to go
to the project menu he must wait all the speech music1 music2 and music 3
if there is any way to go to project menu during the speech
thanks and regards
|
This is an actual dialplan application that I wrote. It's a "spike" -- a
proof of concept that is all depth and no breadth. It's known to work in
Asterisk 1.8.
The sound files "ajs_juke01" and "ajs_anykey" you will need to create for
yourself, depending what MP3s you have available (and replace ajs_ with your
own prefix). You can interrupt the announcements or the MP3s by pressing keys
while playing.
;;;;;;;;;;; VERY PRIMITIVE JUKE BOX CONTEXT ;;;;;;;;;;;
[vpjb]
exten => s,1,Background(ajs_juke01)
; "Press 1 for Ocean Colour Scene, 2 for Crowded House"
exten => s,n,WaitExten(1)
exten => s,n,Goto(1)
exten => i,1,Hangup()
exten => 1,1,Background(ajs_anykey)
; "Press any key to stop the music and return to the menu"
exten => 1,n,MP3Player(/songs/09_policemen+pirates.mp3)
exten => 1,n,Goto(vpjb,s,1)
exten => 2,1,Background(ajs_anykey)
; "Press any key to stop the music and return to the menu"
exten => 2,n,MP3Player(/songs/15_distant_sun.mp3)
exten => 2,n,Goto(vpjb,s,1)
exten => _X,1,Hangup()
--
AJS
Answers come *after* questions.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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murf at parsetree.com
Guest
|
| Posted: Thu Nov 28, 2013 10:57 am Post subject: [asterisk-users] issue with speech in IVR |
|
|
On Thu, Nov 28, 2013 at 8:36 AM, Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)> wrote:
| Quote: | hi i follow your dialplan but the issue still the same ican't stop the speech and go to another contextÂ
any other idea  pleaseÂ
best regards .
|
​My guess is that your DTMF tones are not reaching Asterisk. Seen it many times.
Study the path whereby the DTMF is generated and recognized and processed by
Asterisk. What kind of device are you using? Dahdi? SIP? You can use the
rtp set debug to see if the DTMF is coming thru; look at your channel config,
there may be something there that might prevent DTMF. Same with the phone settings.
Best of Luck,
murf​
Â
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dott com ([email]murf at parsetree dott com[/email])
☎ 307-899-5535 |
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salah.elharit200 at gm...
Guest
|
| Posted: Thu Nov 28, 2013 11:37 am Post subject: [asterisk-users] issue with speech in IVR |
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thanks steve for your response i use dahdi. and  in my sip.conf i have dtmfmode=auto
idon't know if i must to put relaxdtmf=yes ? in sip.conf or i need to it in another files
FYI i have a diguim card with dahdi and asterisk 1.4Â
thanks and regardsÂ
2013/11/28 Steve Murphy <murf@parsetree.com (murf@parsetree.com)>
| Quote: |
On Thu, Nov 28, 2013 at 8:36 AM, Salaheddine Elharit <salah.elharit200@gmail.com (salah.elharit200@gmail.com)> wrote:
| Quote: | hi i follow your dialplan but the issue still the same ican't stop the speech and go to another contextÂ
any other idea  pleaseÂ
best regards .
|
​My guess is that your DTMF tones are not reaching Asterisk. Seen it many times.
Study the path whereby the DTMF is generated and recognized and processed by
Asterisk. What kind of device are you using? Dahdi? SIP? You can use the
rtp set debug to see if the DTMF is coming thru; look at your channel config,
there may be something there that might prevent DTMF. Same with the phone settings.
Best of Luck,
murf​
Â
--
Steve Murphy
ParseTree Corporation
57 Lane 17
Cody, WY 82414
✉ murf at parsetree dott com ([email]murf+at+parsetree+dott+com[/email])
☎ 307-899-5535
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
        http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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paul.belanger at polyb...
Guest
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| Posted: Thu Nov 28, 2013 8:55 pm Post subject: [asterisk-users] issue with speech in IVR |
|
|
On 13-11-28 05:22 AM, Salaheddine Elharit wrote:
| Quote: | hello,
i have add the the code below but the issue still the same i can't go to
the project during the speech
any other solution
best regards
NB:for the version of asterisk i can't move to another version for the
moment
exten => _X,1,NoOp(Digit entered during prompt)
exten => _X,2,Goto(project,s,1)
|
Then you have a DTMF issue, Background will allow DTMF to interrupt the
prompts.
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter:
https://twitter.com/pabelanger
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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asterisk_list at earth...
Guest
|
| Posted: Thu Nov 28, 2013 11:05 pm Post subject: [asterisk-users] issue with speech in IVR |
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|
On 28/11/13 15:36, Salaheddine Elharit wrote:
| Quote: | hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context
any other idea please
best regards .
|
It sounds as thgough the DTMF tones are not being sent in a way that Asterisk is seeing .....
What type of telephone technology are you using? Hardware SIP phones, software SIP phones, analogue phones via an FXS card, analogue phones via a SIP ATA? What codec are you using?
If you make an extension-to-extension call, can you send DTMF tones down the line? Both ways around? Do they decode properly? (You can get a mobile phone app for this.)
--
AJS
Answers come *after* questions. |
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salah.elharit200 at gm...
Guest
|
| Posted: Fri Nov 29, 2013 3:05 am Post subject: [asterisk-users] issue with speech in IVR |
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hi
yes if imake an extension-to-extension call, i can send DTMF, Both ways ==== yes
in my case i don't need a Hardware SIP phone or a software SIP phones
i have just a number 05xxxxxx600
when the customer call this number i stor his number in my database and i call him later
if he press 1 for xxxxxx 1 press 2 for yyyyyyy
i sotre his phone number and his choice in my database
for me the issue the customer he can nto wait the speech of unless xxxx and yyyy finished .
best regards
i use a diguim card with PRI
2013/11/29 A J Stiles <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)>
| Quote: | On 28/11/13 15:36, Salaheddine Elharit wrote:
| Quote: | hi i follow your dialplan but the issue still the same ican't stop the speech and go to another context
any other idea please
best regards .
|
It sounds as thgough the DTMF tones are not being sent in a way that Asterisk is seeing .....
What type of telephone technology are you using? Hardware SIP phones, software SIP phones, analogue phones via an FXS card, analogue phones via a SIP ATA? What codec are you using?
If you make an extension-to-extension call, can you send DTMF tones down the line? Both ways around? Do they decode properly? (You can get a mobile phone app for this.)
--
AJS
Answers come *after* questions.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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isrlgb at gmail.com
Guest
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| Posted: Fri Nov 29, 2013 3:12 am Post subject: [asterisk-users] issue with speech in IVR |
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Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use the keypad to control the playback
-----Original Message-----
From: Salaheddine Elharit <salah.elharit200@gmail.com>
Sender: asterisk-users-bounces@lists.digium.comDate: Fri, 29 Nov 2013 08:05:16
To: Asterisk Users Mailing List - Non-Commercial Discussion<asterisk-users@lists.digium.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Subject: Re: [asterisk-users] issue with speech in IVR
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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http://www.asterisk.org/hello
asterisk-users mailing list
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_____________________________________________________________________
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mitul at enterux.in
Guest
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| Posted: Fri Nov 29, 2013 3:50 am Post subject: [asterisk-users] issue with speech in IVR |
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Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode = auto
Mitul On Nov 29, 2013 1:42 PM, <isrlgb@gmail.com (isrlgb@gmail.com)> wrote: |
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asterisk_list at earth...
Guest
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| Posted: Fri Nov 29, 2013 5:02 am Post subject: [asterisk-users] issue with speech in IVR |
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On Thursday 28 November 2013, Salaheddine Elharit wrote:
| Quote: | hi
i follow your dialplan but the issue still the same ican't stop the speech
and go to another context
any other idea please
best regards .
|
Well, the Background() application should definitely allow you to interrupt a
sound being played.
One possibility is that DTMF tones being generated within a call are not
reaching Asterisk. If you phone someone and then press keys in-call, do they
hear the DTMF tones in their handset?
What kind of phones are you using? Analogue phones via an ATA, SIP phones or
something else?
Have you a scrap PC on which you could temporarily install a newer Asterisk
and Dahdi version and try my VPJB app again? (I don't actually think your
problem is caused by an out-of-date version, but it wouldn't hurt to check.)
--
AJS
Answers come *after* questions.
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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salah.elharit200 at gm...
Guest
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| Posted: Fri Nov 29, 2013 9:23 am Post subject: [asterisk-users] issue with speech in IVR |
|
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hello
i add the following in chan_dahdi and the issue has been solved thanks a lot for your help and support now ican stop the speech and go to my context
i really appreciate your help and support
2013/11/29 Mitul Limbani <mitul@enterux.in (mitul@enterux.in)>
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