Hello,
I have been reviewing people posts about trunking between servers and can not seem to find anything that applies to my issue.
I have two servers one in CALGARY and one in VANCOUVER. The servers talk just fine when I dial by extensions. However it is a lack of understanding of call flow over IAX that I think is my issue, I am sure i have a syntax issue. The way I want it to work is if I dial any number beginning with 9403 the call will route via the calgary server and same for Vancouver. IE:
IF 9403XXXXXXX Then pass call to -> CALGARY for outward local dialing.
IF 9604XXXXXXX Then pass call to -> VANCOUVER for outward local dial.
Right now I have the following config:
VANCOUVER########################
–IAX.CONF–
[size=75][CALGARY]
Type = friend
user = ######
secret = #######
host = ###.###.###.###[/size]
–EXTENSIONS.CONF–
[size=75][numberplan-custom-1]
exten = _9403XXXXXXX.,1,Dial(IAX2/CALGARY/${EXTEN:1})[/size]
CALGARY#########################
–IAX.CONF–
[size=75][KAMLOOPS]
Type = friend
User = ######
Secret = #######
Host = ###.###.###.###[/size]
–EXTENSIONS.CONF–
[size=75][numberplan-custom-1]
exten = _9604XXXXXXX.,1,Dial(IAX2/VANCOUVER/${EXTEN:1})[/size]
According to the calgary server the call does not get that far when initiated from Vancouver
ASTERISK DATA FOUND ON VANCOUVER SERVER:
[size=75][b] – Executing [94037######@numberplan-custom-1:1] Macro(“SIP/3000-08214ba8”, “trunkdial|/4037######”) in new stack
– Executing [s@macro-trunkdial:1] Dial(“SIP/3000-08214ba8”, “/4037######”) in new stack
[Jan 3 17:58:08] WARNING[3866]: channel.c:3188 ast_request: No channel type registered for ‘’
[Jan 3 17:58:08] WARNING[3866]: app_dial.c:1106 dial_exec_full: Unable to create channel of type ‘’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [s@macro-trunkdial:2] Goto(“SIP/3000-08214ba8”, “s-CHANUNAVAIL|1”) in new stack
– Goto (macro-trunkdial,s-CHANUNAVAIL,1)
– Executing [s-CHANUNAVAIL@macro-trunkdial:1] NoOp(“SIP/3000-08214ba8”, “”) in new stack
== Auto fallthrough, channel ‘SIP/3000-08214ba8’ status is ‘CHANUNAVAIL’
<— Transmitting (NAT) to 192.168.3.102:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.3.102:5060;branch=z9hG4bK2827395313036815394;received=192.168.3.102;rport=5060
From: 3000 sip:3000@192.168.3.203;tag=2461524258
To: “94037######” sip:94037######@192.168.3.203;tag=as3677af43
Call-ID: 280191617-58193121781@192.168.3.102
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:94037######@192.168.3.203
Content-Length: 0
X-Asterisk-HangupCause: Channel not implemented
X-Asterisk-HangupCauseCode: 66[/b][/size]
IF I am mistaken about another post on this topic you have my appologies, I just cant seem to find anything…
Thank you
-Jim Akeson