asterisk users - Nov 2005 - How to make write and read formats equal to native format?

I am playing around with different codecs between 2 * servers. However I
don't seem to have any impact on bandwith.
I always get something like this:


           Name: IAX2/ds02-1
           Type: IAX2
       UniqueID: 1131421484.2
      Caller ID: s
 Caller ID Name: (N/A)
    DNID Digits: (N/A)
          State: Up (6)
          Rings: 0
   NativeFormat: 2
    WriteFormat: 64
     ReadFormat: 64
1st File Descriptor: -1
      Frames in: 182
     Frames out: 468
 Time to Hangup: 0
   Elapsed Time: N/A
  Direct Bridge: Zap/1-1
Indirect Bridge: Zap/1-1
 --   PBX   --
        Context: foo
      Extension: s
       Priority: 1
     Call Group: 0
   Pickup Group: 0
    Application: Bridged Call
           Data: Zap/1-1
    Blocking in: ast_waitfor_nandfds
      Variables:
BRIDGEPEER=Zap/1-1
DIALEDPEERNUMBER=ds02/95

Native format represents what I've chosen for codec of preference. However,
data exchange is done with Read/Write Format 64 (16 bit Signed Linear PCM).


Any idea on how to enforce native format into read and write streams?

Any help appreciated.

Regards,
B.

Branko Samardzic wrote:
> Any idea on how to enforce native format into read and write streams?
In the peer definition (iax.conf or sip.conf) put:

disallow=all
allow=CODEC_YOU_WANT

where CODEC_YOU_WANT is something like gsm, g729, ulaw etc (keep it to one
entry at both ends and you're guaranteed they'll use it).

-- 
Cheers,

Matt Riddell
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Hi,
I am trying to make following setup

PSTN --> Asterisk_1 ------- IAX2 stream over Internet -----> Asterisk_2
-->
PSTN
             ^
             |
          AGI control App

Basically this is calling card solution that accepts calls from PSTN of one
country
checks user credentials (PIN and access rights), collects destination number
and
establishes IAX2 connection with another Asterix server in different country
that
is used to terminate call on another PSTN.
So far I made everything working nice except one VERY important thing:

IAX2 stream over Internet is at least in one direction bearing 16 bit Signed
Linear PCM
data although I insisted on GSM codec. As link has limited bandwith (64kpbs)
it becomes
congested with single call.
My iax.conf files contain disallow=all and allow=gsm in general section on
both servers
but it is of no help.
Is there any chance to tweak IAX2 stream to be symetric (i.e. to bring GSM
encoded audio
in both directions) and how to achieve that.
Any help is appreciated.
Regards,
B.S.