Codec translation issue

Asterisk Asterisk Support
1

Hi,

I’ve been tinkering with Asterisk on an embedded device and I ran into an issue I haven’t seen before.

The output of ‘core show translation’:

[code]*CLI> core show translation
Translation times between formats (in milliseconds) for one second of data
Source Format (Rows) Destination Format (Columns)

      g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
 g723    -   -    -    -        -     -    -     -    -     -    -    -    -
  gsm    -   -    -    -        -     -    -     -    -     -    -    -    -
 ulaw    -   -    -    -        -     -    1     -    -     -    -    -    -
 alaw    -   -    -    -        -     -    -     -    -     -    -    -    -

g726aal2 - - - - - - - - - - - - -
adpcm - - - - - - - - - - - - -
slin - - 1 - - - - - - - - - -
lpc10 - - - - - - - - - - - - -
g729 - - - - - - - - - - - - -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - - - - - - - - - - - - -
g722 - - - - - - - - - - - - -
*CLI>
[/code]

Since this is an embedded device I’m using precompiled packages (Asterisk 1.4.11).

ulaw, gsm, etc. all work, but the only sound files I’ve loaded due to space limitations are gsm. So when a call is made and it attempts to play a gsm file, the above error happens.

Is this the result of the options used when compiling? Or is this a setting somewhere?

2

To add some info, I’ve also seen this error when a call is made in one format (for example, ulaw) and voicemail format is set to another (such as wav49). It appears that somehow all transcoding capabilities are disabled.

Since it’s a fundamental part of Asterisk to handle transcoding between unlike codecs I wasn’t aware it was even possible to disable this.

Has anyone else seen this?

3

just drop all the codec_* files from another asterisk install into /usr/lib/asterisk/modules

4

Hi

When using * on an embedded device its normal not to have transcoding as the CPU “normally” wont have the grunt to do it.

Ian