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@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@mutualphone
> -- SIP/mutualphone-6b26 is ringing
> -- SIP/mutualphone-6b26 answered IAX2/iaxrene@iaxrene/2
>
> The BT101 gives this:
>
> -- Called 003165193XXXX@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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