[asterisk-users] SayDigits playback doesn't always work

Danny Nicholas danny at debsinc.com
Mon Jan 16 09:26:13 CST 2012


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-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Roland
Sent: Monday, January 16, 2012 9:22 AM
To: asterisk-users at 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 <xiphias at 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 at StumpelZwaag:1]
Wait("SIP/000B822FD265-0000003d", "2") in new stack

[Jan 16 15:54:17]     -- Executing [123 at StumpelZwaag:2]
SayDigits("SIP/000B822FD265-0000003d", "123") in new stack

[Jan 16 15:54:17]     -- <SIP/000B822FD265-0000003d> Playing 'digits/1.gsm'
(language 'nl')

[Jan 16 15:54:17]     -- <SIP/000B822FD265-0000003d> Playing 'digits/2.gsm'
(language 'nl')

[Jan 16 15:54:18]     -- <SIP/000B822FD265-0000003d> Playing 'digits/3.gsm'
(language 'nl')

[Jan 16 15:54:18]     -- Auto fallthrough, channel
'SIP/000B822FD265-0000003d' 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 at StumpelZwaag:1]
Answer("SIP/000B822FD265-0000003e", "") in new stack

[Jan 16 15:54:29]     -- Executing [200 at StumpelZwaag:2]
BackGround("SIP/000B822FD265-0000003e", "main-menu") in new stack

[Jan 16 15:54:29]     -- <SIP/000B822FD265-0000003e> Playing 'main-menu.gsm'
(language 'nl')

[Jan 16 15:54:30]     -- Executing [200 at StumpelZwaag:3]
WaitExten("SIP/000B822FD265-0000003e", "5") in new stack

[Jan 16 15:54:34]   == CDR updated on SIP/000B822FD265-0000003e

[Jan 16 15:54:34]     -- Executing [123 at StumpelZwaag:1]
Wait("SIP/000B822FD265-0000003e", "2") in new stack

[Jan 16 15:54:36]     -- Executing [123 at StumpelZwaag:2]
SayDigits("SIP/000B822FD265-0000003e", "123") in new stack

[Jan 16 15:54:36]     -- <SIP/000B822FD265-0000003e> Playing 'digits/1.gsm'
(language 'nl')

[Jan 16 15:54:36]     -- <SIP/000B822FD265-0000003e> Playing 'digits/2.gsm'
(language 'nl')

[Jan 16 15:54:37]     -- <SIP/000B822FD265-0000003e> Playing 'digits/3.gsm'
(language 'nl')

[Jan 16 15:54:37]     -- Auto fallthrough, channel
'SIP/000B822FD265-0000003e' 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

 

 

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