A colleague called me through my * system via FWD using SJPhone and the quality was distinctly poor - a lot of hum and delay. Looking at the debug log the codec used was miscellaneously numbered 0, 4 and 8. I thought I'd disabled 4 (g.723) but it appears not. My sip.conf has this:
general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = voip-sip defaultexpiry = 3600 register => 12345:[EMAIL PROTECTED]/39 disallow=all allow=alaw allow=ulaw
I was expecting this would stop g.723 from being even tried - am I missing something?
Is there any config option for SJphone that clobbers g.723?
Iain _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
