Defect #2601
closed
Outbound dialing works for everything except 3333/4444
Added by Grant L over 10 years ago.
Updated over 10 years ago.
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!
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
- Status changed from New to In progress
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.
- Status changed from In progress to To be closed
Great! Good that it is working now. I will close the issue.
- Status changed from To be closed to Closed
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