Defect #2601
closed
Outbound dialing works for everything except 3333/4444
0%
Description
I can dial out to PSTN without any issues, but can't seem to dial 3333 or 4444 (I receive all circuits are busy message)
Asterisk Log
Executing [s@macro-dialout-trunk:23] NoOp("IAX2/xxx-xxxx", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 127") in new stack
Server logs show loopback 482 error.
CDRTool SIP trace SIP session 50361f5f0ca3b8232e68cd492a68a00e@sip2sip.info
Any idea why all calling seems to work except for the two test numbers? Thanks!
Updated by Tijmen de Mes over 10 years ago
Then INVITE you send is wrong. It arrives on 'our' node11, yet it is:
INVITE sip:3333@85.17.186.7 SIP/2.0
Via: SIP/2.0/UDP 108.82.17.151:5060;branch=z9hG4bK53662a1b;report
Note the IP in the line: 85.17.186.7, this should be sip2sip.info. In this way what happens is:
- Packet arrives on a random node
- Gets routed to your home node
- Gets routed to the destination, since the domain is not local (it is an ip)
- Packet arrives on node which is the target
- Gets routed back to home node, hence we have loop.
So the fix is make the invite look right like: INVITE sip:3333@sip2sip.info SIP/2.0
Updated by Tijmen de Mes over 10 years ago
- Status changed from New to In progress
Updated by Grant L over 10 years ago
Thanks for the info, I changed HOST=sip2sip.info&81.23.228.129&81.23.228.150&85.17.186.7 in my trunk settings to HOST=sip2sip.info and I can now make test calls.
Updated by Tijmen de Mes over 10 years ago
- Status changed from In progress to To be closed
Great! Good that it is working now. I will close the issue.
Updated by Tijmen de Mes over 10 years ago
- Status changed from To be closed to Closed