i seem to be running into this a lot and changing it on the fly no longer works
[May 1 11:33:48] WARNING[24489][C-00000057]: channel.c:5309 set_format: Unable to find a codec translation path from (g729) to (gsm|ulaw)
[May 1 11:33:48] WARNING[24489][C-00000057]: file.c:1017 ast_streamfile: Unable to open vm-login (format (g729)): No such file or directory
[May 1 11:33:48] WARNING[24489][C-00000057]: app_voicemail.c:10512 vm_authenticate: Couldn’t stream login file
how do i deal with this?
in my sip.conf i have the following
disallow = all ; Good practice dictates disallowing codecs first, and then allowing only the ones we want
allow = g729 ; 8kHz at 8kbps - NOTE: This codec should not generally be enabled without installing Digium’s G.729 transcoding module for Asterisk
allow = ulaw ; 8kHz at 64kbps, North America
allow = alaw ; 8kHz at 64kbps, Worldwide
allow = g722 ; 16kHz at 64kbps
allow = g726 ; 8kHz at 32kbps
allow = slin ; 8kHz at 128kbps - NOTE: This codec should generally not be used outside of a LAN
allow = slin16 ; 16kHz at 256kbps - NOTE: This codec should generally not be used outside of a LAN
Thank you !
Looks like you have neither a valid G.729 licence, nor pre-converted G.729 sound files.
do i need the license - can i use a different codec
You probably don’t need it; most equipment can speak more than just G.729 as a codec.
Remove the allow= line that mentions G.729, reload SIP, and try again.
That seems to have done it - thank you.
Now…what is the advantage of this codec - i am sure there is a nice marketing post somewhere - but in reality do i need it with digium phones ?
A listing of codecs supported by Digium’s phones is here:
wiki.asterisk.org/wiki/display/ … figuration
Your options are:
G.722, a 16kHz codec that consumes 64kbps
G.711 ulaw, an 8kHz codec used in the US, Canada and Japan, that consumes 64kbps.
G.711 alaw, an 8kHz codec used worldwide, that consumes 64kbps.
G.726, an 8kHz codec that consumes 32kbps
G.729a, an 8kHz codec that consumes 8kbps (but requires a commercial module on the Asterisk side at $10 USD / channel - see store.digium.com/productview.php … 8G729CODEC )
Signed linear at 8kHz consuming 128kbps (not a good idea outside of a LAN)
Signed linear at 16kHz consuming 256kbps (not a good idea outside of a LAN)
ok - since we are PRI / analog lines - seems that we do not need this codec for the time being - thank you for the link and the info
i will keep this in mind for future sip implementations
If you don’t have network bandwidth problems, you should only enable one codec, and that should be mu-law or a-law, depending on where you are in the world. That is the codec that will be used on the PSTN.
The main use of G.729 is when you have limited VoIP bandwidth, and are only interested in speech.
The hgh rate codecs are mainly for internal calls between executives in the sort of company were executives get lots of perks. (There may rare cases where there is a valid business case, but never where one party is using a normal PSTN phone.)
that’s good to know
All my “sip” is internal and once it goes out its going to either a POTS or PRI
Thanks for the tip!