Re: [asterisk-users] How to tell what codec is used for each end of a call MD110-H323-SIP
[EMAIL PROTECTED] wrote: Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk for SIP I use 'sip show channels' I'm not sure what the equivilent h323 command is. And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Try 'rtp debug' and the rtp packets should scroll by. Thank you for your time and effort to respond. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to tell what codec is used for each end of a call MD110-H323-SIP
Hi. Calling from Ericsson MD110 via H.323 trunk to an asterisk 1.4.4 I get the call established but no sound heard on either end. What is the best/correct way to try and see what codecs Asterisk is using on each end of the call as it passes through Asterisk? And is there any way to see that voice is in fact being passed through Asterisk during the call (some counters etc.)? Thank you for your time and effort to respond. Baldvin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users