SIP trunk can't receive calls

Asterisk Asterisk Support
1

I’m sure there is a configuration mismatch, I setup my trunk to my SIP provider but it is still messed up and I can make outbound calls but I cannot receive any :frowning:

here is an output of the console set to verbose 10:

-- Executing NoOp("SIP/1xxx4828987-083866a8", "Received incoming SIP connection from unknown peer to 200") in new stack
-- Executing Set("SIP/1xxx4828987-083866a8", "DID=200") in new stack
-- Executing Goto("SIP/1xxx4828987-083866a8", "s|1") in new stack
-- Goto (from-sip-external,s,1)
-- Executing Ringing("SIP/1xxx4828987-083866a8", "") in new stack
-- Executing GotoIf("SIP/1xxx4828987-083866a8", "0?from-trunk|200|1") in new stack
-- Executing Set("SIP/1xxx4828987-083866a8", "TIMEOUT(absolute)=15") in new stack
-- Channel will hangup at 2007-01-06 18:12:36 UTC.
-- Executing Answer("SIP/1xxx4828987-083866a8", "") in new stack
-- Executing Wait("SIP/1xxx4828987-083866a8", "2") in new stack
-- Executing Playback("SIP/1xxx4828987-083866a8", "ss-noservice") in new stack
-- Playing 'ss-noservice' (language 'en')
-- Executing PlayTones("SIP/1xxx4828987-083866a8", "congestion") in new stack
-- Executing Congestion("SIP/1xxx4828987-083866a8", "5") in new stack

== Spawn extension (from-sip-external, s, 8) exited non-zero on ‘SIP/1xxx4828987-083866a8’
– Executing NoOp(“SIP/1xxx4828987-083866a8”, “Hangup”) in new stack
– Executing Set(“SIP/1xxx4828987-083866a8”, “DID=s”) in new stack
– Executing Goto(“SIP/1xxx4828987-083866a8”, “s|1”) in new stack
– Goto (from-sip-external,s,1)
– Executing Ringing(“SIP/1xxx4828987-083866a8”, “”) in new stack
– Executing GotoIf(“SIP/1xxx4828987-083866a8”, “0?from-trunk|s|1”) in new stack
– Executing Set(“SIP/1xxx4828987-083866a8”, “TIMEOUT(absolute)=15”) in new stack
– Channel will hangup at 2007-01-06 18:12:48 UTC.
– Executing Answer(“SIP/1xxx4828987-083866a8”, “”) in new stack
– Executing Wait(“SIP/1xxx4828987-083866a8”, “2”) in new stack
– Executing Playback(“SIP/1xxx4828987-083866a8”, “ss-noservice”) in new stack
– Playing ‘ss-noservice’ (language ‘en’)
– Executing PlayTones(“SIP/1xxx4828987-083866a8”, “congestion”) in new stack
– Executing Congestion(“SIP/1xxx4828987-083866a8”, “5”) in new stack
== Spawn extension (from-sip-external, s, 8) exited non-zero on ‘SIP/1xxx4828987-083866a8’

maybe someone seems what the heck is going on. its supposed to go to extension 200 as my incoming route is stated.

2

Make sure that you have the host= setting correct on the trunk. It seems as if Asterisk thinks that the call is not coming in on the defined trunk.

3

hmm. well i put host=sip.babytel.ca as they had stated in their configuration example. unfortunately they dont really provide support for it and i don’t see what else it should be set to?

4

Have you got a register= line in sip.conf?

5

register=1xxx4828987:password@sip.babytel.ca:5065

thats the line I have. they use port 5065 instead of 5060

6

I have a similar problem with sipgate.co.uk. I’ll try to sort it and let you know what I find.