Since a few days, we have a strange issue with all our inbound routes (4 total). I call the number and FreePBX plays the “noservice.slin16” message that the number is not in service or unavailable. The message tells the number which is the username of the trunk and not the number I was calling.
I have had no issue with this FreePBX installation for months and we haven’t changed anything. Strange is also, out of 4 tries, it works 2 times, and 2 times it plays the noservice message. The call always reaches FreePBX. I see the FreePBX output with Putty, so I rule out issues with the provider.
Looks like some issues with inbound routes.
Any hint is greatly appreciated.
Is it possible that the format of the incoming number changed so that the inbound routes don’t work anymore?
Could you check the log or CDR to see if the numbers match?
Thanks - the caller ID is set to ANY for all inbound routes. I compared the DID in the inbound routes with the CDR report and the format looks okay too.
However, I think that I found a lead. I see in the CDR report that every time noanswer plays, there is no DID in the CDR reports. Perhaps that is a hint that the provider doesn’t always submit the DID with an incoming call.
Create an inbound route with neither a DID nor a CID defined, then those calls should get caught.
It’s an approach to resolve. However I need to route the calls. We have different parties.
It’s hard to route calls without any routing information (particularly DID or CID) available.
Perhaps there are other SIP headers that have that info?
As @dicko says. At the Asterisk command prompt, type
pjsip set logger on
or if the trunk in question is using chan_sip
sip set debug on
wait for a failing call and look at the incoming INVITE in the Asterisk log. With luck, it will have the DID in the To header (or possibly a different header).
Just a hint, many (especially newbies) find
sngrep more intuitive to look at your SIP traffic,just drill down (press enter) to see the ‘headers’ in an “INVITE” transaction.
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