I have had the issue for years. The problem is that Asterisk developers are removed from the business. We desperately need simple way to eliminate transcoding when unnecessary. Transcoding brings a server to its knees. It is a very simple new setting in sip.conf prioritize_matching_codecs=yes I vote for this new feature. However, I don't have the expertise to write a patch. I would say that only Digium developers could attempt to do this without disrupting the code too much. I also tried to migrate to PJSIP, but had to go back when I realized there was no channel variable contaning the inbound IP address. In general, any channel hast to provide the information to the dialplan, somehow, otherwise we cannot do business. I hope the PJSIP integration matures soon.
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
