Chris Maciejewski wrote: > I do have codec_g726 loaded. As I mentioned before > Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite > there is only fpm-sunshine.wav file. It is only MeetMe which is not > working: > > -- <SIP/OpenSER-08208098> Playing 'entering-conf-number.slin' > (language 'en') > [May 22 18:07:04] WARNING[16881]: app_playback.c:447 playback_exec: > ast_streamfile failed on SIP/OpenSER-08208098 for entering-conf-number
This is not MeetMe, it's Playback. You specified a filename with '.slin' in it to Playback, so then Asterisk attempts to find a filename called 'entering-conf-number.slin.<foo>' where <foo> is the possible formats that Asterisk could transcode from. Filenames specified to Playback should not include the format extension. > -- Executing [...@services:7] SayNumber("SIP/OpenSER-08208098", > "1") in new stack > -- <SIP/OpenSER-08208098> Playing 'digits/1.slin' (language 'en') This did not fail. The .slin extension was added by ast_streamfile after it found the correct format to play for this channel. > -- Executing [...@services:8] Wait("SIP/OpenSER-08208098", "1") in new > stack > -- Executing [...@services:9] MeetMe("SIP/OpenSER-08208098", > "11,MI") in new stack > == Parsing '/etc/asterisk/meetme.conf': == Found > -- Created MeetMe conference 1023 for conference '11' > -- <SIP/OpenSER-08208098> Playing 'vm-rec-name.slin' (language 'en') Again, this did not fail. > -- Hungup 'DAHDI/pseudo-1131226973' The only failure of any kind that I see in this log is the call to Playback. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users