Thanks Matt, can I transform the format in dial plan? On Wed, May 11, 2011 at 11:24 AM, Matt Riddell <li...@venturevoip.com>wrote:
> On 11/05/11 3:11 PM, John Wu wrote: > >> Hi Enrico >> thanks I do what u said but meet this problem: >> [May 11 11:08:49] WARNING[4545]: file.c:650 ast_openstream_full: File >> fail.wav does not exist in any format >> [May 11 11:08:49] WARNING[4545]: file.c:953 ast_streamfile: Unable to >> open fail.wav (format 0x2 (gsm)): No such file or directory >> [May 11 11:08:49] WARNING[4545]: app_playback.c:471 playback_exec: >> ast_streamfile failed on SIP/IMSI460020656177633-00000000 for fail.wav >> > > When you playback a file in Asterisk you don't provide the extension. > > So you'd do Playback(fail) rather than Playback(fail.wav) > > 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. > > 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. > > Basically just drop the extension. > > -- > Cheers, > > Matt Riddell > _______________________________________________ > > http://www.venturevoip.com/news.php (Daily Asterisk News) > http://www.venturevoip.com/exchange.php (Full ITSP Solution) > http://www.venturevoip.com/cc.php (Call Centre Solutions) > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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