On 6 Mar 2009, at 19:29, tracinet wrote: > That stinks... We are migrating to SIP from IAX2 at the moment and > running into the same exact problem. No way to control the > destination context unless you use the "fromuser". Of course that > is rendering Caller ID useless as you pointed out.
Give me the exact sip.conf you have both ends. Might be able to get it working. _______________________________________________ -- 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
