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 > [EMAIL PROTECTED] > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]>
_______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
