I setup a asterisk environment with the sip protocols and the codec used is g729. But for some calls, after connect there is no voice in the line.
Asterisk - 1.4.21.
I purchased 151 licenses of g729.
Specification : 64bit
codec_g729a-1.4_3.0.3-nocona.tar.gz
Ouptut of show g729 CLI command.
show g729
2/4 encoders/decoders of 151 licensed channels are currently in use.
Output of show translation CLI command,
show translation
Translation times between formats (in milliseconds) for one second of data
Source format (Row) Destination format (Columns)
g723 gsm ulaw alaw g726aa12 adpcm slin lpc10 g729 speex ilibc g726 g722
g723 - - - - - - - - - - - - -
gsm - - 2 2 2 2 1 4 9 - - 2 -
ulaw - 2 - 1 2 2 1 4 9 - - 2 -
alaw - 2 1 - 2 2 1 4 9 - - 2 -
g7266aa12 - 2 2 2 - 2 1 4 9 - - 1 -
adpcm - 2 2 2 2 - 1 4 9 - - 2 -
slin - 1 1 1 1 1 - 3 8 - - 1 -
lpc10 - 3 3 3 3 3 2 - 10 - - 3 -
g729 - 2 2 2 2 2 1 4 - - - 2 -
speex - - - - - - - - - - - - -
ilibc - - - - - - - - - - - - -
g726 - 2 2 2 1 2 1 4 9 - - - -
g722 - - - - - - - - - - - - -
and the phone which i am using is Cisco 7940 and it support the g729 codec. The problem is 30% of calls are no voice calls. Please Kindly direct me in right direction.