Re: [Asterisk-Users] how to pass G723.1
Kamran Ahmad wrote: hello how to pass G723.1 to other side is there any softphone using g723.1. i want to use G723.1 in my voice communication. Microsoft Netmeeting can use G723.1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No audio when h323 calls are incoming
Did you ever find a solution to this problem? [EMAIL PROTECTED] wrote: All, I have tried very hard to make asterisk work with h323 but still strying: I have been successful making this work SIP -- Asterisk -- h323 -- termination ; But the following: h323 -- asterisk -- h323 -- Termination : works , call set up is ok but then no audio is applied .There is no NAT here at all are public. I also tried h323 -- asterisk -- SIP -- terminatino: I have same problem here, audio I use g723 codec (passthrough ) Can anyone advise what is to look or is it meant not to work anyway ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 codecs
Hi, Can anyone confirm that if I want to do h323 proxying that I do not need codecs installed? For example if the codec being used is g723.1, I don't need the codec installed locally because there is no compression or decompression being done on my server; the incoming traffic is simply being sent out on another h323 channel (h323 in-h323 out). Is this correct? Thanks, Chetan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
