Problems with codec translation when using Monitor and MixMonitor

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Asterisk Users 3 Comments

Hi folks,

I’m having problems when I try to record my calls using MixMonitor or
Monitor. Calls are working well and audio quality is good.
But I just can’t get recorded audio in one leg with both applications. It
happens with internal calls too. As it seems, the problem is my g729
licensing escheme. (just one license installed)

What is the least number of licenses that are needed per recorded call?
or what can I do to fix it?

My asterisk version: Asterisk 1.8.7.1

Logs with MixMonitor:
[Jan 11 17:55:48] WARNING[19500]: translate.c:256
ast_translator_build_path: No translator path from alaw to unknown
[Jan 11 17:55:48] WARNING[19500]: translate.c:256
ast_translator_build_path: No translator path from alaw to unknown

testpbx*CLI> g729 show licenses
0/1 encoders/decoders of 1 licensed channels are currently in use

Licenses Found:
File: G729-… — Key: G729-… — Host-ID: … — Channels: 1 (Expires:
…) (OK)

Logs with Monitor:
[Jan 11 17:49:49] WARNING[19491]: translate.c:256
ast_translator_build_path: No translator path from alaw to g723
[Jan 11 17:49:49] WARNING[19491]: file.c:186 ast_writestream: Unable to
translate to format wav49, source format g729

testpbx*CLI> g729 show licenses
0/1 encoders/decoders of 1 licensed channels are currently in use

Licenses Found:
File: G729-…lic — Key: G729-… — Host-ID: … — Channels: 1 (Expires:
…) (OK)

I’ve searched for this on forums but can’t find a complete answer yet.

Thanks!

Elder D. Arohuanca

Lima – Peru

3 thoughts on - Problems with codec translation when using Monitor and MixMonitor

  • The simplest answer, I purchased one additional license and one
    simultaneous call is being recorded now. I do not understand why the ulaw
    codec (or format) is involved here (… No translator path from alaw to
    unknown …)

    Any entry will be very appreciated.

    Elder D. Arohuanca

    Lima – Peru

  • When you say ‘call’, do you mean a call between two phones (endpoints)?
    If so, and both endpoints are using G.729 for audio, then yes, recording
    that call in any format other than G.729 will require *two* G.729
    decoders, one for each audio stream being received by Asterisk. Even in
    a case where you are only recording the combined audio from the two
    phones (MixMonitor), the audio must still be decoded in order to be mixed.

  • Here is a matrix we put together about g729 license needs:

    ======================== ====================== ========================= ====== ======= ======== ========
    Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders
    ======================== ====================== ========================= ====== ======= ======== ========
    ulaw ulaw yes yes yes 0 0
    ulaw ulaw yes yes no 0 0
    ulaw ulaw yes no no 0 0
    ulaw ulaw yes no yes 0 0

    ulaw ulaw no yes yes 0 0
    ulaw ulaw no yes no 0 0
    ulaw ulaw no no no 0 0
    ulaw ulaw no no yes 0 0

    ulaw g729 yes yes yes 3 3
    ulaw g729 yes yes no 2 3
    ulaw g729 yes no no 1 1
    ulaw g729 yes no yes 3 3

    ulaw g729 no yes yes 3 3
    ulaw g729 no yes no 2 3
    ulaw g729 no no no 1 1
    ulaw g729 no no yes 3 3

    g729 ulaw yes yes yes 2 5
    g729 ulaw yes yes no 2 5
    g729 ulaw yes no no 1 1
    g729 ulaw yes no yes 2 3

    g729 ulaw no yes yes 2 5
    g729 ulaw no yes no 2 5
    g729 ulaw no no no 1 1
    g729 ulaw no no yes 2 3

    g729 g729 yes yes yes 4 7
    g729 g729 yes yes no 3 7
    g729 g729 yes no no 1 1
    g729 g729 yes no yes 4 5

    g729 g729 no yes yes 4 7
    g729 g729 no yes no 3 7
    g729 g729 no no no 1 1
    g729 g729 no no yes 4 5