[Asterisk-Users] No compatible codecs

[jeffrey johnson]

this may help you

http://billing.mutualphone.com/phpBB2/viewtopic.php?t=78&postdays=0&postorder=asc&start=15



On Tue, 18 Jan 2005 10:23:45 -0500, Kanuri, Seshu (Company IT)
<Seshu.Kanuri at morganstanley.com> wrote:
> 
> 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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