password can have it by snooping anyhow. :-) T.
On 1 Jun 2009, at 07:18, Rob Hillis wrote:
The clue in the log is "no authority found". Something in the configuration at the other end doesn't match the configuration at this end - almost certainly the username and password. Why are you including the IP address when dialling the trunk? If your peers are set up with IP addresses (which they are) it should not be necessary.By the way, it's a *very* bad idea to post passwords in a public forum.Tharanga wrote: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/[email protected] /4567,10,t") in new stack-- Called [email protected]/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=XXXXXXXXXXXXXX context=sip,sip2,sip3 ;keyrotate=off permit=0.0.0.0/0.0.0.0 1.6 EXTENSIONS.CONF [globals] TRUNKIAX14=IAX2/[email protected] [sip] ;exten => 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t) exten => 4567,1,Voicemail(${EXTEN},u) ~ 1.2 EXTENSIONS.CONF[Jun 1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_process: Rejected connect attempt from 147.120.203.71, who was trying to reach '4567@[trunk14] type=friend host=147.120.203.71 auth=plaintext secret=Mah context=sip,sip2,sip3 ;keyrotate=off permit=0.0.0.0/0.0.0.0 [globals] TRUNKIAX14=IAX2/[email protected] [sip] exten => s,1,wait(1) ; Answer the line exten => s,n,BackGround(demo-congrats) exten => s,n,ResponseTimeout,5 exten => s,n,Dial(SIP/${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_______________________________________________ -- 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
Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk
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