I’ve got 2 trunks, one is only for DID (didforsale.com), I cannot use it for outbound. The second I use for my outbound termination (flowroute.com).
I set up my outbound route trunk sequence, 0 is flowroute, didforsale isn’t listed at all.
When I attempt to make an outbound call, I get “all circuits are busy now” and my asterisk CLI shows this…
Executing [[email protected]:22] Dial(“SIP/701-00000008”, “SIP/didforsale_did/1612xxxxxxx,300,”) in new stack
I don’t understand why it’s doing anything at all with that trunk!
I’m sure you will need more information, but I’m pretty new to the whole VOIP thing, can anyone get me on the right track?
24 hour bump. Sorry if this is considered improper.
Need system information and versions please. Especially FreePBX version and how it was installed.
Logs would be helpful, a database dump from outbound would be good too.
Thanks for the response.
Installed via PBXinaFlash version 188.8.131.52
FreePBX version 184.108.40.206
asterisk version 220.127.116.11
Running on CentOS 6.2
Here is my log from /var/log/asterisk/full
Can you tell me how to do the database dump?
The custom SIP flowroute is in the wrong context so you are essentially routing the call back you yourseld
how to dump:
select * from outbound_route_patterns_patterns;
select * from outbound_route_sequence;
select * from outbound_route_trunks;
Also peer and user details from trunks.
Once again, thank you for your response.
I’m not sure what you mean by flowroute being in the wrong context… should it not be in from-trunk?
I’m not sure how to paste the results of the SQL queries in a non-eye shattering way, so I’ll just type them in; hopefully this is what you are looking for.
Outbound route patterns under match_pattern_pass.
All have route_id of 7, prefix, cid and prepend all blank:
Outbound route sequence:
route_id 7 seq 0
outbound route trunks:
Route_id 7, trunk_id 2, seq 0
Flowroute peer details:
User details blank
didforsale peer details:
user details blank
register string blank
Can I provide any more information?
I’d really like to get this resolved.
If someone knows of a good paid support, I am willing to go that route.
Go to the top of this page under store.
whew… $150… that’s a little more than I was hoping for since I can’t think this will take nearly an hour once someone looks into it. I’ll have to keep googling I guess.
“The custom SIP flowroute is in the wrong context so you are essentially routing the call back you yourself”
Can you elaborate on this? How do I change the context?
So I found the issue. there was an errant line in my extensions_custom.conf which was forcing extension 701 to use trunk 1. I got rid of the line and everything now works as it should.
eg0e, that’s why I saw it going to the wrong trunk.
Your log didn’t reconcile to your trunks settings so I had nothing else to off er you.