Hi all I hope that someone of you can solve this. Right now I'm stuck!!!!! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182)
All I can see in CLI is: == Using SIP RTP CoS mark 5 -- Executing [182@default:1] Dial("SIP/181-0000000a", "SIP/182") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-0000000b is ringing -- SIP/182-0000000b is making progress passing it to SIP/181-0000000a -- SIP/182-0000000b answered SIP/181-0000000a -- Remotely bridging SIP/181-0000000a and SIP/182-0000000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-0000000a' Seems like extension 182 (called ext) is getting call and passing them another time to me 181 (origin call) I've try it with siemens pbx and works as expected cheers! Sergio -- _____________________________________________________________________ -- 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