[Asterisk-Users] IVR : Can't hear my message

Antoine LOUIS antoine.louis at gmail.com
Thu Apr 6 01:49:35 MST 2006


Hello,

I've reccorded a voice message for the IVR. (.wav, 16 bits, 8kHz)
The file is /var/lib/asterisk/sound/11ivrrecording.wav.

When asterisk (1.2.5) starts this file i can't hear it on my phone.

Here is the log :

Apr  6 17:00:16 VERBOSE[845] logger.c:     -- Executing
SetCallerID("SIP/11-97b9", ""Patrice" <11>") in new stack
Apr  6 17:00:16 VERBOSE[845] logger.c:     -- Executing NoOp("SIP/11-97b9",
"Using CallerID "Patrice" <11>") in new stack
Apr  6 17:00:16 VERBOSE[845] logger.c:     -- Executing
Playback("SIP/11-97b9", "11ivrrecording") in new stack
Apr  6 17:00:16 DEBUG[845] channel.c: Scheduling timer at 160 sample
intervals
Apr  6 17:00:16 VERBOSE[845] logger.c:     -- Playing '11ivrrecording'
(language 'en')
Apr  6 17:00:17 DEBUG[26916] chan_sip.c: Stopping retransmission on '
xqOZotDBq6ICZb9l at 192.168.42.24' of Response 2: Match Found
Apr  6 17:00:49 DEBUG[26916] chan_sip.c: Stopping retransmission on '
4c14706a2d71d234273cdc26207692b1 at 192.168.42.10' of Request 102: Match Found
Apr  6 17:00:50 DEBUG[845] channel.c: Scheduling timer at 0 sample intervals
Apr  6 17:00:50 VERBOSE[845] logger.c:   == Spawn extension (from-internal,
*99, 2) exited non-zero on 'SIP/11-97b9'


Anyone has an idea ?

Thanks a lot.

Antoine
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060406/a34e115f/attachment.htm


More information about the asterisk-users mailing list