Not my idea - just what I came across on google - probably should open a JIRA issue so it gets "really resolved" instead of hit-and-miss patching.
-----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: Saturday, December 03, 2011 2:38 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] video calls not working On 11-11-21 10:07 AM, Danny Nicholas wrote: > Two items > > #1 you only need 1 disallow=all in your sip.conf definition > > #2 you need to patch rtp.c to define 126 as FORMAT_H263 - this is an > xlite response to Asterisk starting music-on-hold during the connect > pause. The r on the dial command attempts to do a "faux ring" which > xlite interprets as a MOH request, so if you don't want to > patch/recompile, just take the r off of Dial. > Why are you manually patching asterisk? Have you created an issue in JIRA about this? -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- 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 -- _____________________________________________________________________ -- 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