voice service voip sip session transport tcp Last I checked, asterisk doesn't support TCP SIP signaling (or RTP over TCP). See what happens if you change it back to the UDP default.
On 12/7/06, FaberK <[EMAIL PROTECTED]> wrote:
Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- "I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated!" _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
