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
Maybe Matching 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)
