[asterisk-users] Can't get G.726 to work.

[Kevin P. Fleming]

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 [501 at 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 [501 at services:8] Wait("SIP/OpenSER-08208098", "1") in new stack
>     -- Executing [501 at 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: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org