That’s an interesting question and there is a logical approach.
Asterisk will not load a module without a valid configuration file. These configuration are stored in /etc/asterisk
If you view the log file of Asterisk /var/log/asterisk/full and try and load the SIP module from the CLI (or simply restart asterisk) you would see a complaint about the missing sip.conf.
At some point everything got out of sync. The amportal script is the main control of FreePBX and performs sanity check along with restarting apps. It does not cold start Asterisk, sometime ‘service asterisk stop’ count to 10 ‘service asterisk start’ is the only way to shake things loose.
SIP registration is “broken” for me, too. I suspected a firewall issue, but I’ve been through everything in this thread, including amportal stop/start, and I’m still having the same issue. I’m using AsteriskNOW 1.5, downloaded yesterday. I am using Cisco 7960 phones with SIP firmware.
One extension from my sip_additional.conf:
[6021]
deny=
type=friend
secret=********
qualify=yes
port=5060
pickupgroup=
permit=
nat=yes
mailbox=6021@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/6021
context=from-internal
canreinvite=no
callgroup=
callerid=device <6021>
allow=ulaw
accountcode=
call-limit=50
First line from console when I try and dial another extension:
– Executing [6021@from-sip-external:1] NoOp(“SIP/192.168.1.28-087071c8”, “Received incoming SIP connection from unknown peer to 6021”) in new stack
Results of sip show peers:
Name/username Host Dyn Nat ACL Port Status
6022 (Unspecified) D N 0 UNKNOWN
6021 (Unspecified) D N 0 UNKNOWN
2 sip peers [Monitored: 0 online, 2 offline Unmonitored: 0 online, 0 offline]
I set nat=no. What makes me question if SIP is working properly, is when I run tail on /var/log/asterisk/full, I don’t see any SIP handshakes when I power on the phone–only once I try and place a call from the phones.