Hi, I'm new to Asterisk and SIP, but have experience with developing VoIP apps in general and H.323 apps in particular.
I would like to fix a problem we're having with Asterisk performing codec conversion in the following situation: EP1, preferred codec order aLaw, G.729 EP2, preferred codec order G.729 EP1 places call to EP2, we see two call legs: EP1 to * is aLaw * to EP2 is G.729 (so Asterisk is performing codec conversion) In this scenario we'd like to use G.729 for both call legs with the media stream bypassing Asterisk. Thus reducing the CPU load on the * machine and the need for additional G.729 licenses. We would like Asterisk to use the codec settings of EP2 while sending a reinvite to EP1. Currently Asterisk uses it's own codec settings as defined in sip.conf. Since we are using Asterisk together with SER, our endpoints are not defined in sip.conf so we cannot put information about them in it. Can someone tell me where to start developing? Which files, functions? TIA. -- Andreas Sikkema Rits tele.com Scheepmakersstraat 11 3011 VH Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 2245540 _______________________________________________ Asterisk-Dev mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
