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 -- Executing [...@services:7] SayNumber("SIP/OpenSER-08208098", "1") in new stack -- <SIP/OpenSER-08208098> Playing 'digits/1.slin' (language 'en') -- 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') -- Hungup 'DAHDI/pseudo-1131226973' 2009/5/22 Kevin P. Fleming <kpflem...@digium.com>: > Chris Maciejewski wrote: >> Yes, I was missing "allow=g726" for this peer :-( >> >> Playback(/var/lib/asterisk/moh/fpm-sunshine) >> >> works OK now, however I still can't get MeetMe to work. >> >> Before I had similar problem, when MeetMe wasn't working with GSM >> codec because I was missing .gsm audio files. >> I suspect now it is the same problem, as I don't have audio files for G726? >> >> Will try converting .pcm to .g726 and see if that will fix MeetMe issue. > > If you have codec_g726 loaded, you should be able to use prompt files in > any format that Asterisk can transcode from/to. 'core show translations' > should show you what formats Asterisk can convert to and from G.726. > > -- > 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 > _______________________________________________ -- 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