my sip phone registered on 1.6, when i dial 4567 from 1.6 version, it wont go to 1.6 voice mail. it says
== Using SIP RTP CoS mark 5 -- Executing [4...@sip:1] Dial("SIP/312-09f9a720", "IAX2/trun...@147.120.203.98/4567,10,t") in new stack -- Called trun...@147.120.203.98/4567 [Jun 1 11:01:18] WARNING[8178]: chan_iax2.c:8991 socket_process: Call rejected by 147.120.203.98: No authority found -- Hungup 'IAX2/trunk14-9738' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/312-09f9a720' status is 'CHANUNAVAIL' [trunk14] type=friend host=147.120.203.98 auth=plaintext secret=Mah context=sip,sip2,sip3 ;keyrotate=off permit=0.0.0.0/0.0.0.0 1.6 EXTENSIONS.CONF [globals] TRUNKIAX14=IAX2/trun...@147.120.203.98 [sip] ;exten => 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t) exten => 4567,1,Voicemail(${EXTEN},u) ~ 1.2 EXTENSIONS.CONF [Jun 1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_process: Rejected connect attempt from 147.120.203.71, who was trying to reach '4567@ [trunk14] type=friend host=147.120.203.71 auth=plaintext secret=Mah context=sip,sip2,sip3 ;keyrotate=off permit=0.0.0.0/0.0.0.0 [globals] TRUNKIAX14=IAX2/trun...@147.120.203.71 [sip] exten => s,1,wait(1) ; Answer the line exten => s,n,BackGround(demo-congrats) exten => s,n,ResponseTimeout,5 exten => s,n,Dial(SIP/${EXTEN},20,t) ;exten => s,n,BackGround(goodbye) exten => s,n,Hangup exten => 4567,1,Dial(${TRUNKIAX14}/${EXTEN},10,t) Asterisk versions may differ. I do IAX trunk successfully even between Asterisk 1.0.2 and 1.4.xx please post your Dial command. _______________________________________________ -- 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