I cant figure it out but why dont you dont you make an extension that you can dial and record yoursef. Exten => 100,1,Wait(2) Exten => 100,2,Record(FileName:gsm) Exten => 100,3,Wait(2) Exten => 100,4,Playback(FileName) Exten => 100,5,Hangup Antoine LOUIS <antoine.louis@gmail.com> wrote: 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@192.168.42.24 ' of Response 2: Match Found Apr 6 17:00:49 DEBUG[26916] chan_sip.c: Stopping retransmission on '4c14706a2d71d234273cdc26207692b1@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 _______________________________________________ --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 --------------------------------- Blab-away for as little as 1?/min. Make PC-to-Phone Calls using Yahoo! Messenger with Voice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060407/f21efc5b/attachment.htm