Unable to leave voice mail

Asterisk Asterisk Support
1

Hello,

Unable to leave voice mail, getting this…

[Jan 26 21:38:15] WARNING[3009]: chan_sip.c:3751 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x4 (ulaw)(4)/0x2 (gsm)(2)
[Jan 26 21:38:15] WARNING[3009]: chan_sip.c:3751 sip_write: Asked to transmit frame type 4, while native formats is 0x100 (g729)(256) read/write = 0x4 (ulaw)(4)/0x2 (gsm)(2)
[Jan 26 21:38:15] WARNING[3009]: channel.c:2930 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm)
[Jan 26 21:38:15] WARNING[3009]: file.c:147 ast_stopstream: Unable to restore format back to 2
– Recording the message
– x=0, open writing: /var/spool/asterisk/voicemail/default/1000/tmp/KTVIpv format: wav49, 0x81f9d58
– x=1, open writing: /var/spool/asterisk/voicemail/default/1000/tmp/KTVIpv format: gsm, 0x81fa038
– x=2, open writing: /var/spool/asterisk/voicemail/default/1000/tmp/KTVIpv format: wav, 0x81f9748
[Jan 26 21:38:15] WARNING[3009]: channel.c:2930 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x40 (slin)
[Jan 26 21:38:15] WARNING[3009]: app.c:589 __ast_play_and_record: Unable to set to linear mode, giving up
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on ‘SIP/64.154.41.100-081efeb8’ in macro ‘stdexten’
== Spawn extension (default, 1000, 1) exited non-zero on 'SIP/64.154.41.100-081efeb8’
sip*CLI>

I do have g.729, ulaw, gsm allow.

Any help will be appreciated

Regards

AL

2

Hi,

I think you are on to it. Look at the codecs.

At the CLI> show translation what does it say.

3

i think you don’t have g.729 this is a codec that need a licence! if you don’t have then you need to setup all you phone’s and system settings ith other codecs like ulaw or alaw

4

Thx, here is the information…

sip*CLI> 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    -   -    2    2        2     2    1     4    -     -    -    2    -
 ulaw    -   3    -    1        2     2    1     4    -     -    -    2    -
 alaw    -   3    1    -        2     2    1     4    -     -    -    2    -

g726aal2 - 3 2 2 - 2 1 4 - - - 1 -
adpcm - 3 2 2 2 - 1 4 - - - 2 -
slin - 2 1 1 1 1 - 3 - - - 1 -
lpc10 - 4 3 3 3 3 2 - - - - 3 -
g729 - - - - - - - - - - - - -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 - 3 2 2 1 2 1 4 - - - - -
g722 - - - - - - - - - - - - -
sip*CLI>

But my question why asterisk try to save voice mail files on 729 codec?

Regars,

Al

5

Hi,

You don’t have g729 codec installed you need to buy a license from Digium but you should be able to use another codec.

  1. What is in your voicemail.conf under general.

[general]
format=wav49|gsm|wav

  1. In your SIP.conf (assuming your are using a sip phone) disallow it
    disallow=g729

  2. On your phone disable or make it the lowest choice.

Brett

6

Some points:

(a) you will be able to talk with g729 through asterisk without a license.

(b) If you want to listen to music on hold or IVR announcements in another format other than g729, you will need to have a license (for each concurrent channel) to transcode between the g729 audio, and the files on your phone system.

© If you leave voicemail, the same applies - transcoding from g729 to the recorded format.

(a) is called pass-thru, and no license is required.

(b) + © you need a license. This is not an asterisk thing but whoever came up with the g729 codec put some patent/license on it.

7

Issue solved. Conf file corrupted, Had to re intall Asterisk, working working

Regards to all,

AL :smiley: