Thanks Steve, See my answers inline > -----Original Message----- > From: Steve Kann [mailto:[EMAIL PROTECTED] > Sent: Tuesday, December 21, 2004 1:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] codec issues > > Shoval Tomer wrote: > > >We've bought the G729 codec for lowering SIP bandwidth usage (we're > >using grandstream phones) and we're quite happy with it up until I tried > >using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. > > > >Weirdly enough, calls from IAXphone to the GS phone work just fine. > >So are calls from both phones to voicemail. And from both phones to > >analog phones connected to FXS ports. > > > >Calls from GS to IAXphone ring, and once I answer the call in IAXphone, > >I hear a very load noise. > > > > > On which side do you hear this noise? IAXphone, or GS? >
I hear the noise on IAX site > Does the noise continue, or just go for a few milliseconds? Continue till I hang up > > Does your version of IAXphone support multiple codecs, or just GSM? Supports only GSM > > >Asterisk CLI shows this: > >channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/205/1 > >of format GSM since our native format has changed to G729A > > > >(not just once, over and over and over again till I hang up) > > > >my sip.conf entry for the grandstream phone shows > >disallow=all > >allow=g729 > >and > >reinvite=no > > > >I did 'iax2 show channels' and 'sip show channels' > > > >When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and > >the Sip channel shows codec G729A > > > >When I call the other way around, Sip shows G729A and IAX2 shows GSM. > > > >Hmm, seems ok... > > > >I tried changing my sip conf to include allow=g729,gsm > > > >Now the calls sounds fine, but the bandwidth is uses is near 20K instead > >of just 6K (both phones are near me, and the Asterisk server is at a > >remote location, and I can monitor bandwidth usage in my FW). > > > >Can anyone help? > > > > > > It obviosuly sounds like codec negotion, on one side or another, isn't > working, and you're sending an incompatible codec to the other side, or > the other side doesn't know what codec is being sent.. > How can I even control that? And sip show channels and iax2 show channels show the correct codecs. > Using ethereal to see what's happening on the network would show you > what's going on pretty clearly.. > > -SteveK > I'll try but it's not going to be easily done, as the asterisk server is at a remote location and I'm no ethereal expert... > > > > _______________________________________________ > Asterisk-Users mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support. _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
