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

[Roland]

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
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120116/099a4a5b/attachment.htm>