Hi Karl, that's funny you are asking this, am also currently looking at how to solve the g722 codec negotiation riddle, in my particular case to play nicely together with a KonfTel 300 IP conference phone.
> In other words, incoming calls are easy since codecs are negotiated > from least-known (the remote party) to most-known (my endpoint) and my > codecs can simply be preferred accordingly to match the remote. Look at setting the channel variable_SIP_CODEC - however it might or might not work depending on your version of Asterisk, see for example: https://issues.asterisk.org/view.php?id=13243 Here's a dialplan snippet that might give you another hint or two. exten => 123,1,NoOp(-- Inbound read: ${CHANNEL(audioreadformat)} --) exten => 123,n,NoOp(-- Inbound native: ${CHANNEL(audionativeformat)} --) exten => 123,n,Set(WIDEBAND=0) exten => 123,n,Set(WIDEBAND=${REGEX("g722" ${SIPPEER(${SIPCHANINFO(peername)}:codecs)})}) exten => 123,n,ExecIf($[${WIDEBAND} = 1]|Set|_SIP_CODEC=g722) exten => 123,n,Dial(SIP/abc123) Please note the SPACE between ${REGEX("g722" and ${SIPPEER > Outbound calls seem harder. Our endpoints always negotiate g.722 between > themselves and Asterisk and then Asterisk must transcode to the preferred > codec of the REMOTE party. Not ideal. Together with the g722 transcoding patch for Asterisk 1.4.17 it does not work out, unfortunately. Currently I cannot make a statement on a more recent 1.4 release. g722 patch: http://users.netplex.net/~andrew/asterisk/#g722 Older patch that I use for 1.4.17: http://users.netplex.net/~andrew/asterisk/g722-20080110.patch.gz However I can successfully employ setting _SIP_CODEC if in the example above instead of "Dial()" I do a "MusicOnHold()" - both with or without a preceeding ANSWER; without means early audio playing of the native g722 encoded MOH file. My snom starts out with alaw, and then we switch to g722. > Is there an elegant way to do this? Consider the codec negotiation patch? I'd be interested to hear about your results! http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch https://issues.asterisk.org/view.php?id=4825 Philipp -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
