Thanks Matt,<br>can I transform the format in dial plan?<br><br><div class="gmail_quote">On Wed, May 11, 2011 at 11:24 AM, Matt Riddell <span dir="ltr"><<a href="mailto:lists@venturevoip.com" target="_blank">lists@venturevoip.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;"><div>On 11/05/11 3:11 PM, John Wu wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
Hi Enrico<br>
thanks I do what u said but meet this problem:<br>
[May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File<br>
fail.wav does not exist in any format<br>
[May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to<br>
open fail.wav (format 0x2 (gsm)): No such file or directory<br>
[May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec:<br>
ast_streamfile failed on SIP/IMSI460020656177633-00000000 for fail.wav<br>
</blockquote>
<br></div>
When you playback a file in Asterisk you don't provide the extension.<br>
<br>
So you'd do Playback(fail) rather than Playback(fail.wav)<br>
<br>
That way if you have 10 files all called fail (i.e. fail.wav, fail.gsm, fail.ulaw, fail.alaw etc etc) it will pick the one which matches the phone calls.<br>
<br>
For example in the above example you were making a call in the GSM format but it couldn't find a file called fail.wav.gsm so it looked for fail.wav.* and couldn't find anything.<br>
<br>
Basically just drop the extension.<br>
<br>
-- <br>
Cheers,<br>
<br>
Matt Riddell<br>
_______________________________________________<br>
<br>
<a href="http://www.venturevoip.com/news.php" target="_blank">http://www.venturevoip.com/news.php</a> (Daily Asterisk News)<br>
<a href="http://www.venturevoip.com/exchange.php" target="_blank">http://www.venturevoip.com/exchange.php</a> (Full ITSP Solution)<br>
<a href="http://www.venturevoip.com/cc.php" target="_blank">http://www.venturevoip.com/cc.php</a> (Call Centre Solutions)<div><div></div><div><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br>