Inbound route issue

Hello,

I have just set up a new system using Teliax as the trunk provider. I can call between extensions and I can make outbound calls. However, when I make a call to my Teliax DID, I receive the “The number you have dialed is not in service…” message. I know the call is making it to my system because the Asterisk log shows that it received the call and that it’s going to play the above message. I tried to change the destination to the phone book or the ‘congestion’ sound, but I still get the aforementioned message. Here’s what the log looks like:

[Dec 1 16:59:06] VERBOSE[18582] logger.c: – Executing [[email protected]:1] GotoIf(“SIP/TeliaxPeer-1b357320”, “0?from-trunk||1”) in new stack
[Dec 1 16:59:06] VERBOSE[18582] logger.c: – Executing [[email protected]:2] Set(“SIP/TeliaxPeer-1b357320”, “TIMEOUT(absolute)=15”) in new stack
[Dec 1 16:59:06] VERBOSE[18582] logger.c: – Channel will hangup at 2009-12-01 21:59:21 UTC.
[Dec 1 16:59:06] VERBOSE[18582] logger.c: – Executing [[email protected]:3] Answer(“SIP/TeliaxPeer-1b357320”, “”) in new stack
[Dec 1 16:59:06] VERBOSE[18582] logger.c: – Executing [[email protected]:4] Wait(“SIP/TeliaxPeer-1b357320”, “2”) in new stack
[Dec 1 16:59:08] VERBOSE[18582] logger.c: – Executing [[email protected]:5] Playback(“SIP/TeliaxPeer-1b357320”, “ss-noservice”) in new stack
[Dec 1 16:59:08] VERBOSE[18582] logger.c: – <SIP/TeliaxPeer-1b357320> Playing ‘ss-noservice’ (language ‘en’)
[Dec 1 16:59:14] VERBOSE[18582] logger.c: – Executing [[email protected]:6] PlayTones(“SIP/TeliaxPeer-1b357320”, “congestion”) in new stack
[Dec 1 16:59:14] VERBOSE[18582] logger.c: – Executing [[email protected]:7] Congestion(“SIP/TeliaxPeer-1b357320”, “5”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: == Spawn extension (from-sip-external, s, 7) exited non-zero on ‘SIP/TeliaxPeer-1b357320’
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:1] NoOp(“SIP/TeliaxPeer-1b357320”, “Hangup”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:2] Set(“SIP/TeliaxPeer-1b357320”, “DID=s”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:3] Goto(“SIP/TeliaxPeer-1b357320”, “s|1”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Goto (from-sip-external,s,1)
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:1] GotoIf(“SIP/TeliaxPeer-1b357320”, “0?from-trunk|s|1”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:2] Set(“SIP/TeliaxPeer-1b357320”, “TIMEOUT(absolute)=15”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Channel will hangup at 2009-12-01 21:59:31 UTC.
[Dec 1 16:59:16] VERBOSE[18582] logger.c: – Executing [[email protected]:3] Answer(“SIP/TeliaxPeer-1b357320”, “”) in new stack
[Dec 1 16:59:16] VERBOSE[18582] logger.c: == Spawn extension (from-sip-external, s, 3) exited non-zero on ‘SIP/TeliaxPeer-1b357320’

Any thoughts?

Thanks!

Pat

I found the answer in another thread.

Under the ‘General Tab’ make sure you have “Allow Anonymous SIP Calls” set to ‘Yes’ under “Security”