asterisk users - Jun 2009 - IAX2 trunking with Older Asterisk, version ?

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 at sip:1] Dial("SIP/312-09f9a720",
"IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack
    -- Called trunk10 at 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/trunk10 at 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/trunk10 at 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.

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 [4567 at sip:1] Dial("SIP/312-09f9a720",
"IAX2/trunk10 at 147.120.203.98/4567,10,t") in new stack
>     -- Called trunk10 at 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=XXXXXXXXXXXXXX
> context=sip,sip2,sip3
> ;keyrotate=off
> permit=0.0.0.0/0.0.0.0
>
>
>
> 1.6 EXTENSIONS.CONF
>
> [globals]
> TRUNKIAX14=IAX2/trunk10 at 147.120.203.98
>
>
> [sip]
> ;exten => 4567,1,Dial(${TRUNKIAX14}/${EXTEN}|10|t)
> exten => 4567,1,Voicemail(${EXTEN},u)
> ~
>
>
>
> 1.2 EXTENSIONS.CONF
>
> [Jun  1 05:20:31] NOTICE[9536]: chan_iax2.c:8782 socket_process: Rejected
connect attempt from 147.120.203.71, who was trying to reach '4567@
>
>
> [trunk14]
> type=friend
> host=147.120.203.71
> auth=plaintext
> secret=Mah
> context=sip,sip2,sip3
> ;keyrotate=off
> permit=0.0.0.0/0.0.0.0
>
>
>
>
>
> [globals]
> TRUNKIAX14=IAX2/trunk10 at 147.120.203.71
>
>
> [sip]
> exten => s,1,wait(1)                     ; Answer the line
> exten => s,n,BackGround(demo-congrats)
> exten => s,n,ResponseTimeout,5
> exten => s,n,Dial(SIP/${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.
>
>
>
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