asterisk users - Aug 2003 - FWD SIP phone format=2, FWD call format=4, why?

Hi!

I'm trying an asterisk-FWD connection.  I'm using X-Lite OR SIPPS as the
IP phone.  I configured the X-Lite and SIPPS to use GSM codec.  Whe I 
call FWD, I get this info on the channels when the call has not been 
stablished yet:

sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  
Format
192.246.69.223   613         1770bf3430d  00102/00000  00000ms  0000ms  2
150.187.xxx.yyy    ildefonso   C72ACD25-1A  00101/11482  00000ms  0000ms  2
2 active SIP channel(s)
    -- SIP/fwd-161b answered SIP/ildefonso-d2fc
    -- Attempting native bridge of SIP/ildefonso-d2fc and SIP/fwd-161b

When it gets stablished, I get:

sip show channels
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  
Format
192.246.69.223   613         1770bf3430d  00102/00000  00000ms  0000ms  4
150.187.xxx.yyy   ildefonso   C72ACD25-1A  00101/11482  00000ms  0000ms  2
2 active SIP channel(s)

My guess: Format 4=G711u, Format 2=gsm.

My question: Is there any way to force SIP to use a codec.  See, we have 
a 1024kbps connection for data and voice, and I don't like the idea of 
"eating" 64kbps of the channel for each call.  Addionaly, when there
are
other people (here we have around 1500 computers, all of them trying to 
get throug the 1024kbps link) using the data link, it gets almost 
imposible to use the voice, unless I put all the other people *VERY* 
slow (using a traffic administrator).

Thanks in advance for your help,

Sincerely,

Ildefonso Camargo
icamargo@unet.edu.ve

Last I looked, FWD is G711 only, unless you use the lite service, then
it is G729 only. No asterisk work will change FWD's setup.
 
On Wed, 2003-08-13 at 09:56, Jose Ildefonso Camargo Tolosa
wrote:
> Hi!
> 
> I'm trying an asterisk-FWD connection.  I'm using X-Lite OR SIPPS
as the
> IP phone.  I configured the X-Lite and SIPPS to use GSM codec.  Whe I 
> call FWD, I get this info on the channels when the call has not been 
> stablished yet:
> 
> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  
> Format
> 192.246.69.223   613         1770bf3430d  00102/00000  00000ms  0000ms  2
> 150.187.xxx.yyy    ildefonso   C72ACD25-1A  00101/11482  00000ms  0000ms  2
> 2 active SIP channel(s)
>     -- SIP/fwd-161b answered SIP/ildefonso-d2fc
>     -- Attempting native bridge of SIP/ildefonso-d2fc and SIP/fwd-161b
> 
> When it gets stablished, I get:
> 
> sip show channels
> Peer             User/ANR    Call ID      Seq (Tx/Rx)  Lag      Jitter  
> Format
> 192.246.69.223   613         1770bf3430d  00102/00000  00000ms  0000ms  4
> 150.187.xxx.yyy   ildefonso   C72ACD25-1A  00101/11482  00000ms  0000ms  2
> 2 active SIP channel(s)
> 
> My guess: Format 4=G711u, Format 2=gsm.
> 
> My question: Is there any way to force SIP to use a codec.  See, we have 
> a 1024kbps connection for data and voice, and I don't like the idea of 
> "eating" 64kbps of the channel for each call.  Addionaly, when
there are
> other people (here we have around 1500 computers, all of them trying to 
> get throug the 1024kbps link) using the data link, it gets almost 
> imposible to use the voice, unless I put all the other people *VERY* 
> slow (using a traffic administrator).
> 
> Thanks in advance for your help,
> 
> Sincerely,
> 
> Ildefonso Camargo
> icamargo@unet.edu.ve
> 
> 
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-- 
Steven Critchfield  <critch@basesys.com>