John Taylor
2008-Nov-27 11:09 UTC
[asterisk-users] trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. "sip show peers" shows my phones registered OK, but the peer describing my SIP trunk does not even display: "sip show peers" Name/username Host Dyn Nat ACL Port Status 204/204 192.168.xxx.xxx D 2048 Unmonitored 203/203 192.168.xxx.xxx D 2048 Unmonitored "sip show registry" sip.voipfone.co.uk:5060 xxxxxxxx 45 Registered Thu, 27 Nov 2008 11:01:56:03 "sip reload" or restarting asterisk with /etc/init.d/asterisk restart fixes the problem and I get the following output: Name/username Host Dyn Nat ACL Port Status 204/204 192.168.xxx.xxx D 2048 Unmonitored 203/203 192.168.xxx.xxx D 2048 Unmonitored voipfone/xxxxxxxx 195.189.173.10 5060 OK (61 ms) "sip show registry" sip.voipfone.co.uk:5060 xxxxxxxx 45 Registered Thu, 27 Nov 2008 11:05:28:02 sip.conf entry for the trunk [voipfone] type=friend secret=xxxxxx username=xxxxxxxx fromuser=xxxxxxxx fromdomain=sip.voipfone.co.uk host=sip.voipfone.co.uk insecure=very dtmfmode=rfc2833 context=fromvoipfone ;inbound calls falls in this context of dialplan disallow=all allow=ilbc ;allow=ulaw ;allow=alaw qualify=yes Any ideas warmly welcomed! Setting debug to level 9 isn't helping me out on this. John
Possibly Parallel Threads
- call queuing not detecting caller hang up when call originates from voip provider
- SIP with Qualify and NAT
- forward call back up same trunk to external cell phone problem
- caller getting cut off intermittently
- asterisk@home inbound call problem to SIP trunk. (voipfone UK)