[asterisk-users] SayDigits
Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards -- _ -- 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
Re: [asterisk-users] SayDigits
HI IS THERE POSIBLE TO MONITOR THE DIGIUM PORTS CHANNEL THROUGH SNMP. IF PASSIBLE MEANS KINDLY SHARE THE SNMP CONFIGURATION OR DOCUMENT FOR THAT. Regards Thangaraj 9994828285 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko Sent: Friday, February 08, 2013 4:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SayDigits Hello Is there a way to slow down or speed up the speed at which SayDigits rattles off a series of digits? Reagards -- _ -- 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 --- E-mail Disclaimer : The information contained herein (including any accompanying documents) is confidential and is intended solely for the addressee(s). If you have erroneously received this message, please immediately delete it and notify the sender. Also, if you are not the intended recipient, you are hereby notified that any disclosure, copying, distribution or taking any action in reliance on the contents of this message or any accompanying document is strictly prohibited and is unlawful. The organization is not responsible for any damage caused by a virus or alteration of the e-mail by a third party or otherwise. The contents of this message may not necessarily represent the views or policies of Sun Business Solutions Pvt Ltd. --- -- _ -- 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
Re: [asterisk-users] SayDigits
IS THERE POSIBLE TO MONITOR THE DIGIUM PORTS CHANNEL THROUGH SNMP Please don't hyjack a thread, start a new message. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] SayDigits
Is there a way to slow down or speed up the speed at which SayDigits core show application saydigits [Synopsis] Say Digits. [Description] This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. [Syntax] SayDigits(digits) [Arguments] Not available So, I'd have to say no. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- 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
Re: [asterisk-users] SayDigits
Am 08.02.2013 13:11, schrieb Doug Lytle: Is there a way to slow down or speed up the speed at which SayDigits core show application saydigits [Synopsis] Say Digits. [Description] This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. [Syntax] SayDigits(digits) [Arguments] Not available So, I'd have to say no. Doug You should write a little AGI-Script instead. -- _ -- 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
Re: [asterisk-users] SayDigits
On 8/2/13 12:11 pm, Doug Lytle wrote: Is there a way to slow down or speed up the speed at which SayDigits So, I'd have to say no. I suppose potentially you could re-record the sound files to 'say' each digit faster (and with shorter rolloff at the end of each word), then put those into a separate [language] folder in /var/lib/asterisk/sounds, then use those instead in your dialplan. You might even be able to process the existing recordings using your favourite audio editing tool to speed the sound files and reduce the rolloff at the end. No guarantees it'll sound any good, mind. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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
Re: [asterisk-users] SayDigits
Hello,
My final solution:
...
same = n,Gosub(dati,s,1(${card}))
[dati]
exten = s,1,NoOp
same = n,Set(say=${LEN(${ARG1})})
same = n,Set(digit=0)
same = n,While($[${digit} ${say}])
same = n,Saydigits(${ARG1:${digit}:1})
same = n,Wait(.75)
same = n,Set(digit=$[${digit} + 1])
same = n,Endwhile
same = n,Return
Thank you for yours suggestion
regards
--
_
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Re: [asterisk-users] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via
a menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are
being played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing
'main-menu.gsm' (language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
with asterisk
;nat=yes; assume the
device is behind nat
secret=xxx ; a secure password for this device
dtmfmode=auto ; accept touch-tones from
devices, negotiated automatically
disallow=all; reset with voice codecs
to accept from, and request to, the device
allow=alaw ; which audio codecs we
accept from
canreinvite=nonat
--
_
-- 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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Re: [asterisk-users] SayDigits playback doesn't always work
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before the
SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but it
should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via a
menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are being
played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm'
(language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
with asterisk
;nat=yes; assume the device
is behind nat
secret=xxx ; a secure password for this device
dtmfmode=auto ; accept touch-tones from
devices, negotiated automatically
disallow=all; reset with voice codecs to
accept from, and request to, the device
allow=alaw ; which audio codecs we
accept from
canreinvite=nonat
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
Re: [asterisk-users] SayDigits playback doesn't always work
Ok, got it. Indeed, starting with Answer() helped.
But I still don't understand why the parking feature isn't working then. I
used the sample config. Transfer the call to 700, playback of the lot is
being executed, but I hear nothing. Probably the same problem, but how do I
change this?
On Mon, Jan 16, 2012 at 4:26 PM, Danny Nicholas da...@debsinc.com wrote:
You aren’t “opening the line” in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are
“plunging” into the SayDigits() function without opening the channel. Some
functions will generate their own Answer() if not present, others will not.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Roland
*Sent:* Monday, January 16, 2012 9:22 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] SayDigits playback doesn't always work
** **
In addition: I tried adding Playback(hello) to the 123 extension, before
the SayDigits. Then everything is being played perfectly.
** **
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but
it should be played back to the one who is parking of course.
** **
So something seems to be wrong with SayDigits?
** **
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:***
*
Hi,
** **
I have this wierd problem where SayDigits does work when I execute it via
a menu, but not when calling directly. In my extensions, I have this setup:
** **
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
** **
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
** **
** **
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
** **
When I call 123 from the same phone, I do see that the sound files are
being played to me, but I don't hear any sound.
** **
In Asterisk CLI I see this:
** **
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
** **
This is the call that doesn't work. Then when I call 200, I see this:
** **
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing
'main-menu.gsm' (language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/1.gsm' (language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing
'digits/2.gsm' (language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing
'digits/3.gsm' (language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
** **
This call works perfectly. What am I missing?
** **
In my sip.conf I have:
** **
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic
Re: [asterisk-users] SayDigits playback doesn't always work
Post your dialplan snippet you use to park the call.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
Ok, got it. Indeed, starting with Answer() helped.
But I still don't understand why the parking feature isn't working then. I
used the sample config. Transfer the call to 700, playback of the lot is
being executed, but I hear nothing. Probably the same problem, but how do I
change this?
On Mon, Jan 16, 2012 at 4:26 PM, Danny Nicholas da...@debsinc.com wrote:
You aren't opening the line in the 123 call. In the 200 call, the
Answer() opens the output audio channel. In the 123 call you are plunging
into the SayDigits() function without opening the channel. Some functions
will generate their own Answer() if not present, others will not.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:22 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] SayDigits playback doesn't always work
In addition: I tried adding Playback(hello) to the 123 extension, before the
SayDigits. Then everything is being played perfectly.
Also when I park a call to 700, I cannot hear the playback of the parking
lot. I do see this in the logs though, so I can pickup the call then, but it
should be played back to the one who is parking of course.
So something seems to be wrong with SayDigits?
On Mon, Jan 16, 2012 at 4:02 PM, Rolandow xiph...@rolandow.com wrote:
Hi,
I have this wierd problem where SayDigits does work when I execute it via a
menu, but not when calling directly. In my extensions, I have this setup:
exten = 200,1,Answer()
same = n,Background(main-menu)
same = n,WaitExten(5)
exten = 123,1,Wait(2)
same = n,SayDigits(${EXTEN})
Now when I call 200, I hear the menu, and then when I press 123, it plays
back one two three. Everything is OK.
When I call 123 from the same phone, I do see that the sound files are being
played to me, but I don't hear any sound.
In Asterisk CLI I see this:
[Jan 16 15:54:15] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:15] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003d, 2) in new stack
[Jan 16 15:54:17] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003d, 123) in new stack
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:17] -- SIP/000B822FD265-003d Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:18] -- SIP/000B822FD265-003d Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:18] -- Auto fallthrough, channel
'SIP/000B822FD265-003d' status is 'UNKNOWN'
[Jan 16 15:54:18] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This is the call that doesn't work. Then when I call 200, I see this:
[Jan 16 15:54:29] == Using SIP RTP CoS mark 5
[Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse
for Notify User 001565150F04.1
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1]
Answer(SIP/000B822FD265-003e, ) in new stack
[Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2]
BackGround(SIP/000B822FD265-003e, main-menu) in new stack
[Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm'
(language 'nl')
[Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3]
WaitExten(SIP/000B822FD265-003e, 5) in new stack
[Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e
[Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1]
Wait(SIP/000B822FD265-003e, 2) in new stack
[Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2]
SayDigits(SIP/000B822FD265-003e, 123) in new stack
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm'
(language 'nl')
[Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm'
(language 'nl')
[Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm'
(language 'nl')
[Jan 16 15:54:37] -- Auto fallthrough, channel
'SIP/000B822FD265-003e' status is 'UNKNOWN'
[Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle
for Notify User 001565150F04.1
This call works perfectly. What am I missing?
In my sip.conf I have:
[stumpel-zwaag](!) ; create template for our
devices
type=friend ; the channel driver will
mathc on username first, IP second
context=StumpelZwaag; this is where calls from
the device will enter the dialplan
host=dynamic; the device will register
Re: [asterisk-users] SayDigits playback doesn't always work
This symptom usually means you are doing an attended transfer instead of a blind transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roland Sent: Monday, January 16, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SayDigits playback doesn't always work Ok, got it. Indeed, starting with Answer() helped. But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this? This is the call that doesn't work. Then when I call 200, I see this: [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1 [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer(SIP/000B822FD265-003e, ) in new stack [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround(SIP/000B822FD265-003e, main-menu) in new stack [Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm' (language 'nl') [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten(SIP/000B822FD265-003e, 5) in new stack [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait(SIP/000B822FD265-003e, 2) in new stack [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits(SIP/000B822FD265-003e, 123) in new stack [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm' (language 'nl') [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm' (language 'nl') [Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm' (language 'nl') [Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-003e' status is 'UNKNOWN' [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1 This call works perfectly. What am I missing? In my sip.conf I have: [stumpel-zwaag](!) ; create template for our devices type=friend ; the channel driver will mathc on username first, IP second context=StumpelZwaag; this is where calls from the device will enter the dialplan host=dynamic; the device will register with asterisk ;nat=yes; assume the device is behind nat secret=xxx ; a secure password for this device dtmfmode=auto ; accept touch-tones from devices, negotiated automatically disallow=all; reset with voice codecs to accept from, and request to, the device allow=alaw ; which audio codecs we accept from canreinvite=nonat -- _ -- 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 -- _ -- 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
Re: [asterisk-users] SayDigits playback doesn't always work
I am just starting with Asterisk .. I think you are right, I am doing an attended transfer, although I don't exactly understand what that means. I still need to know in what lot I can pickup my call again right? Ok, my config .. (i will leave out the commented stuff, because there's lot of comments in the sample config) [general] parkext = 700 ; What extension to dial to park. Set per parking lot. parkpos = 701-720 ; What extensions to park calls on. (defafult parking lot) context = parkedcalls ; Which context parked calls are in (default parking lot) parkingtime = 300 ; Number of seconds a call can be parked before returning. comebacktoorigin = yes ; Setting this option configures the behavior of call parking when the courtesytone = beep; Sound file to play to when someone picks up a parked call parkedplay = both; Who to play courtesytone to when picking up a parked call. Thanks! On Mon, Jan 16, 2012 at 4:59 PM, Eric Wieling ewiel...@nyigc.com wrote: This symptom usually means you are doing an attended transfer instead of a blind transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Roland Sent: Monday, January 16, 2012 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SayDigits playback doesn't always work Ok, got it. Indeed, starting with Answer() helped. But I still don't understand why the parking feature isn't working then. I used the sample config. Transfer the call to 700, playback of the lot is being executed, but I hear nothing. Probably the same problem, but how do I change this? This is the call that doesn't work. Then when I call 200, I see this: [Jan 16 15:54:29] == Using SIP RTP CoS mark 5 [Jan 16 15:54:29] == Extension Changed 137[StumpelZwaag] new state InUse for Notify User 001565150F04.1 [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:1] Answer(SIP/000B822FD265-003e, ) in new stack [Jan 16 15:54:29] -- Executing [200@StumpelZwaag:2] BackGround(SIP/000B822FD265-003e, main-menu) in new stack [Jan 16 15:54:29] -- SIP/000B822FD265-003e Playing 'main-menu.gsm' (language 'nl') [Jan 16 15:54:30] -- Executing [200@StumpelZwaag:3] WaitExten(SIP/000B822FD265-003e, 5) in new stack [Jan 16 15:54:34] == CDR updated on SIP/000B822FD265-003e [Jan 16 15:54:34] -- Executing [123@StumpelZwaag:1] Wait(SIP/000B822FD265-003e, 2) in new stack [Jan 16 15:54:36] -- Executing [123@StumpelZwaag:2] SayDigits(SIP/000B822FD265-003e, 123) in new stack [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/1.gsm' (language 'nl') [Jan 16 15:54:36] -- SIP/000B822FD265-003e Playing 'digits/2.gsm' (language 'nl') [Jan 16 15:54:37] -- SIP/000B822FD265-003e Playing 'digits/3.gsm' (language 'nl') [Jan 16 15:54:37] -- Auto fallthrough, channel 'SIP/000B822FD265-003e' status is 'UNKNOWN' [Jan 16 15:54:37] == Extension Changed 137[StumpelZwaag] new state Idle for Notify User 001565150F04.1 This call works perfectly. What am I missing? In my sip.conf I have: [stumpel-zwaag](!) ; create template for our devices type=friend ; the channel driver will mathc on username first, IP second context=StumpelZwaag; this is where calls from the device will enter the dialplan host=dynamic; the device will register with asterisk ;nat=yes; assume the device is behind nat secret=xxx ; a secure password for this device dtmfmode=auto ; accept touch-tones from devices, negotiated automatically disallow=all; reset with voice codecs to accept from, and request to, the device allow=alaw ; which audio codecs we accept from canreinvite=nonat -- _ -- 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New
[asterisk-users] saydigits in another language
I want to rerecord the 1 2 3 ... 0 sounds, but not overwrite the defaults. So, I've recorded them into a custom directory /var/lib/asterisk/sounds/custom I was hoping to be able to do the following: exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? TIA Julian. ___ --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] saydigits in another language
Julian Lyndon-Smith wrote: however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? 1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] under the main sounds directory (/var/lib/asterisk/sounds/ ???); 2) Also remember to create the same subdirectory under every other main directory (letters, digits, phonetic etc); 3) Copy/move the newly recorded messages into these new directories - numbers into digits. exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) Instead of custom use the ISO code. [1] [1] http://preview.tinyurl.com/btkp ___ --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] saydigits in another language
Not that custom shouldn't work, but you just need to place them in sounds/digits/custom not sounds/custom On 4/15/07, Hermann Wecke [EMAIL PROTECTED] wrote: Julian Lyndon-Smith wrote: however, I get no errors, but still get the default Allison sounds for the digits. Anyone got any clues on what I'm doing wrong ? 1) Create a directory named your_country_iso_code (AR|MX|ES|ETC) [1] under the main sounds directory (/var/lib/asterisk/sounds/ ???); 2) Also remember to create the same subdirectory under every other main directory (letters, digits, phonetic etc); 3) Copy/move the newly recorded messages into these new directories - numbers into digits. exten = foo,1,Set(CHANNEL(language)=custom) exten = foo,2,SayDigits(1234567890) Instead of custom use the ISO code. [1] [1] http://preview.tinyurl.com/btkp ___ --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 ___ --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] saydigits
snip I tried say digits 123 and saydigits 123 both gave no application error /snip 1)its saydigits as in one word and not two 2)As with a lot of functions in asterisk thre data that you are working with has to be in parentheses i.e. Exten = 123,1,Answer Exten = 123,2,Saydigits(1234567890) Exten = 123,3,Hangup __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] saydigits
Jerry Geis wrote:
I was searching on voip-info.org for saydigits.
I see no indication it is not valid in 1.2.4 asterisk.
however, when trying to use it I get and error no application saydigits.
what is the correct way to echo back digits in asterisk 1.2.4?
I tried say digits 123 and saydigits 123 both gave no application
error
Try something like this as an experiment:
; Read back caller's number
exten = 3912,1,Wait(1)
exten = 3912,2,SayDigits(${CALLERID(num)})
exten = 3912,3,Hangup
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[Asterisk-Users] saydigits
I was searching on voip-info.org for saydigits. I see no indication it is not valid in 1.2.4 asterisk. however, when trying to use it I get and error no application saydigits. what is the correct way to echo back digits in asterisk 1.2.4? I tried say digits 123 and saydigits 123 both gave no application error Thanks jerry ___ --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] saydigits
Jerry Geis wrote: I was searching on voip-info.org for saydigits. I see no indication it is not valid in 1.2.4 asterisk. however, when trying to use it I get and error no application saydigits. what is the correct way to echo back digits in asterisk 1.2.4? I tried say digits 123 and saydigits 123 both gave no application error Jerry, I have it on my box: demo*CLI show version Asterisk 1.2.4 built by root @ demo on a i686 running Linux on 2006-02-27 07:15:32 UTC demo*CLI show application saydigits demo*CLI -= Info about application 'SayDigits' =- [Synopsis] Say Digits [Description] SayDigits(digits): This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. See the LANGUAGE function for more information on setting the language for the channel. demo*CLI You might want to check if the application is loaded. ___ --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
[Asterisk-Users] saydigits
Thanks, turns out I wasnt calling the application with parenthis Saydigits(123) is what I needed... THanks, for the help. jerry Jerry Geis wrote: / I was searching on voip-info.org for saydigits. // I see no indication it is not valid in 1.2.4 asterisk. // however, when trying to use it I get and error no application saydigits. // // what is the correct way to echo back digits in asterisk 1.2.4? // // I tried say digits 123 and saydigits 123 both gave no application // error // / Jerry, I have it on my box: demo*CLI show version Asterisk 1.2.4 built by root @ demo on a i686 running Linux on 2006-02-27 07:15:32 UTC demo*CLI show application saydigits demo*CLI -= Info about application 'SayDigits' =- [Synopsis] Say Digits [Description] SayDigits(digits): This application will play the sounds that correspond to the digits of the given number. This will use the language that is currently set for the channel. See the LANGUAGE function for more information on setting the language for the channel. ___ --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
[Asterisk-Users] Saydigits
I had:
exten = 695,2,SayDigits(${CALLERIDNUM}) ; Says your phone number
but it does not work anymore after upgrade. How should it be now?
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[Asterisk-Users] Saydigits
Hi... has anyone written or seen a variation of Saydigits that behaves like Background (listening and responding to DTMF)? If there's such a beast, I'd sure like to know... if not, how hard would it be to implement? Jesus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SayDigits -- ToneDigits??
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the SayDigits(${CALLERIDNUM}) command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
exten = 34,1,Answer
exten = 34,2,Wait(1)
exten = 34,3,Playback(vm-extension)
exten = 34,4,SayDigits(${CALLERIDNUM})
exten = 34,5,Wait(2)
exten = 34,6,Macro(DTMFDigits,${CALLERIDNUM})
exten = 34,7,Hangup
I've searched the voip-info tiki and google, but haven't seen anything
like this mentioned. Can anyone help?
Thanks,
Greg
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Re: [Asterisk-Users] SayDigits -- ToneDigits??
Greg Blakely wrote:
I have a user who wants to receive an ANI spitback in DTMF. Right now,
the SayDigits(${CALLERIDNUM}) command works fine with voice. But I'd
like to end up doing both. Something along the lines of:
CLI show application SendDTMF
-= Info about application 'SendDTMF' =-
[Synopsis]:
Sends arbitrary DTMF digits
[Description]:
SendDTMF(digits[|timeout_ms]): Sends DTMF digits on a channel.
Accepted digits: 0-9, *#abcd
Returns 0 on success or -1 on a hangup.
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[Asterisk-Users] saydigits/background
is there a way to do SayDigits() or equivalent that is
backgrounded?
application is
exten = s,1,Background(zz-fwd-areyouat) ; use callerid or enter
exten = s,2,SayDigits(${CALLERIDNUM}) ; telling callerid
exten = _*,1,Macro(fwd-set,${userid},${CALLERIDNUM})
exten = _*,2,SayDigits(${CALLERIDNUM}) ; if * use callerid
exten = _*,3,Background(zz-fwd-callswillbe) ; report to where calls
exten = _*,4,SayDigits(${EXTEN}); will be forwarded
exten = _*,5,Hangup()
exten = _X.,1,Macro(fwd-set,${userid},${EXTEN})
exten = _X.,2,Background(zz-fwd-callswillbe); if digits use them
exten = _X.,3,SayDigits(${EXTEN}) ; as outbound number
exten = _X.,4,Hangup()
exten = h,1,Hangup()
exten = i,1,GoTo(s,1)
exten = t,1,GoTo(s,1)
[ yes steve, i spent the usual half hour with google and the
wiki ]
randy
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Re: [Asterisk-Users] SayDigits
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/ does not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
ast_streamfile: Unable to open digits/ (format ULAW): No
such file or directory
in Extensions.conf
exten = 1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and there are many
files in there. Any idea's?
Thanks! :)
Chris
have you tried:
exten = 1010,1,SayDigits(${CALLERIDNUM})
?
hth,
greg
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Re: [Asterisk-Users] SayDigits
Awesome, that worked! Thanks :)
Chris
- Original Message -
From: Grzegorz Nosek [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 25, 2004 1:31 AM
Subject: Re: [Asterisk-Users] SayDigits
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
Has anyone had this problem:
(When calling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/ does not exist in any format
Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
ast_streamfile: Unable to open digits/ (format ULAW): No
such file or directory
in Extensions.conf
exten = 1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and there are many
files in there. Any idea's?
Thanks! :)
Chris
have you tried:
exten = 1010,1,SayDigits(${CALLERIDNUM})
?
hth,
greg
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[Asterisk-Users] SayDigits
Has anyone had this problem:
(Whencalling to ext. 1010)
Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
ast_openstream: File digits/" does not exist in any formatJan 24 10:50:27
WARNING[-1252262992]: file.c:734 ast_streamfile: Unable to open digits/" (format
ULAW): No such file or directory
in Extensions.conf
exten =
1010,1,SayDigits(${CALLERID})
/var/lib/asterisk/sounds/digits exists, and
there are many files in there. Any idea's?
Thanks! :)
Chris
Re: [Asterisk-Users] SayDigits
On Tuesday 27 May 2003 20:52, Richard Alexander wrote: I suspect that the (American) voice would have called the hash pound in any case.. :-) Or octothorpe. See http://www.wikipedia.org/wiki/Number%20sign ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
On Tue, 27 May 2003 11:48:19 -0700 (PDT), Brad Bergman wrote: I think SayDigits will say anything for which there is a sound file in the digits directory. So if you put a S.gsm file there, SayDigits,S98 should say Star Nine Eight. I realize that's not exactly what you're looking for. Close, and you are right, anything in the digit directory will work as such, problem i have is * how the heck can you have a asterisk(star).gsm file ??? I notice you use S but thats not really a star. Basically Our system (emulating a normal telco here) uses the star key as a set key and a hash key as a terminator on a string of digits or separator. The hash now works, but the start I need a reback method ( festival sounds terrible) On Tue, 27 May 2003, Gary wrote: Any chance of say digits being extended to recognise * # ?? Heck these are digits on a normal keypad :-) Gary . ___ 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] SayDigits
Ah, now has anyone got a gsm of thevoice for start and hash ?? On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ 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] SayDigits
Heck, I am not that fussy !! Actually, if we could actually get festivel to be fully understandable and using thevoice I think we could all be a lot happier :-) On Tue, 27 May 2003 20:52:58 -0400, Richard Alexander wrote: I suspect that the (American) voice would have called the hash pound in any case.. :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Sent: Tuesday, May 27, 2003 8:38 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SayDigits Ah, now has anyone got a gsm of thevoice for start and hash ?? On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ 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 . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SayDigits
On Tue, 27 May 2003 17:06:30 -0700, Jim Gottlieb wrote: On 2003-05-28 at 09:57, Gary ([EMAIL PROTECTED]) wrote: how the heck can you have a asterisk(star).gsm file ??? I was able to create one with touch \*.gsm so this should work. I doubt asterisk is doing any globbing. ___ we actually can record to file *.gsm #.gsm so thats for that tip. . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
