[Asterisk-Users] No compatible codecs

[Kanuri, Seshu (Company IT)]

Original Post
----------------
I have an Asterisk related problem with mutualphone.
I can connect to any number with any softphone that I am using (iaxcomm,
SJPhone, and a few others).
Also I have a Grandstream BT 101. But I cannot call (via Asterisk) to
mutualphone destinations. Other destinations go fine.

A working phone call (e.g. from iaxcomm) gives the following on the
console:

    -- Accepting AUTHENTICATED call from 192.168.112.99, requested
format = 512, actual format = 512
    -- Called 0031651931985 at mutualphone
    -- SIP/mutualphone-6b26 is ringing
    -- SIP/mutualphone-6b26 answered IAX2/iaxrene at iaxrene/2

The BT101 gives this:

    -- Called 003165193XXXX at mutualphone
    -- SIP/mutualphone-2de1 is ringing
    -- SIP/mutualphone-2de1 answered SIP/chimit01-6013
    -- Attempting native bridge of SIP/chimit01-6013 and
SIP/mutualphone-2de1
Jan 16 18:50:41 WARNING[18631600]: chan_sip.c:2804 process_sdp: No
compatible codecs!
    -- Got SIP response 488 "Not Acceptable Here" back from
209.250.147.116

    show translation (I figure this has anything to do with it) shows
that all paths are supported:

         G723   GSM  ULAW  ALAW  G726 ADPCM SLINR LPC10 G729A SPEEX
ILBC
   G723     -     4     2     2     3     2     1     4    13    35
19
    GSM    15     -     2     2     3     2     1     4    13    35
19
   ULAW    15     4     -     1     3     2     1     4    13    35
19
   ALAW    15     4     1     -     3     2     1     4    13    35
19
   G726    17     6     4     4     -     4     3     6    15    37
21
  ADPCM    15     4     2     2     3     -     1     4    13    35
19
  SLINR    14     3     1     1     2     1     -     3    12    34
18
  LPC10    17     6     4     4     5     4     3     -    15    37
21
  G729A    17     6     4     4     5     4     3     6     -    37
21
  SPEEX    16     5     3     3     4     3     2     5    14     -
20
   ILBC    17     6     4     4     5     4     3     6    15    37
-

The first preferred Vocoder configured in the BT101 is PCMU, but
changing this to G729 (the one that mutualphone is using) won't make it
work. I changed the option back again because all other services (FWD,
BRI, IAX2) work like this and I don't want to break them.

Any suggestions about what I can change to make this work?

Cheers!

Rene Kluwen
Chimit
-----William Suffil's Comment-----
I've heard problems with the Grandstream G729 and the new digium G729 by
MAC ID. Could be a compatibility issue with the implementations.
Did you ever use the Grandstream against asterisk with the old Voiceage
G729? I've heard that works just fine.
-- William

This is not true. I use Grandstream with Digium Codec G729 just fine.
The Old Voiceage codec infact has the problem where the calls do not
connect and when they connect, the quality is horrendous.

My guess is that the entries in SIP.CONF have not been setup properly to
use the available codecs.

Best is to post the SIP.CONF entries here to see what is missing.

By the where did you get the G723 and G729 from? If you have compiled
them on your own, did you statically link the libraries? Or just copied
the .SO files from another dude's Asterisk box?

Post all the details

Seshu Kanuri 
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