Posted: Mon Jun 01, 2009 5:24 am Post subject: [asterisk-users] IAX2 trunking with Older Asterisk, version
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:
Quote:
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 [4567@sip:1] Dial("SIP/312-09f9a720", "IAX2/trunk10@147.120.203.98/4567,10,t") in new stack
-- Called trunk10@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'
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