[Asterisk-Users] FWD SIP phone format=2, FWD call format=4, why?

[Steven Critchfield]

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 at unet.edu.ve
> 
> 
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-- 
Steven Critchfield  <critch at basesys.com>