Hi,
I am toying around with FreePBX 2.5.1.0 with Asterisk 1.4.17 on Ubuntu and at this point I simply want to register a SIP device. So, I went to the “extension” tab and keyed in all the parameters.
In the sip_additional.conf file I see the following after reloading:
[300]
type=friend
secret=100
qualify=yes
port=5060
pickupgroup=1
nat=yes
mailbox=300
host=dynamic
dtmfmode=rfc2833
dial=SIP/300
context=from-internal
canreinvite=no
callgroup=1
callerid=device <300>
accountcode=
call-limit=50
The database show command in the CLI brings up this
DENG*CLI> database show
/AMPUSER/300/cidname : NH
/AMPUSER/300/cidnum : 300
/AMPUSER/300/device : 300
/AMPUSER/300/dictate/email :
/AMPUSER/300/dictate/enabled : enabled
/AMPUSER/300/dictate/format : wav
/AMPUSER/300/language :
/AMPUSER/300/noanswer :
/AMPUSER/300/outboundcid : Nick
/AMPUSER/300/password :
/AMPUSER/300/recording : out=Adhoc|in=Adhoc
/AMPUSER/300/ringtimer : 40
/AMPUSER/300/vmx/busy/state : enabled
/AMPUSER/300/vmx/busy/vmxopts/timeout :
/AMPUSER/300/vmx/unavail/state : enabled
/AMPUSER/300/vmx/unavail/vmxopts/timeout :
/AMPUSER/300/voicemail : default
/CW/300 : ENABLED
/DEVICE/300/default_user : 300
/DEVICE/300/dial : SIP/300
/DEVICE/300/type : fixed
/DEVICE/300/user : 300
/dundi/secret : mDntfVZzMEleCHQxFPawdA==;53HCFchPJSfaDt0nMs6DSA==
/dundi/secretexpiry : 1228339938
I then set up the SIP device correspondingly (user: 300, password: xyz, server: 192.168.0.17) but the phone never registers. A sip debug IP reveals very little unfortunately.
DENG*CLI> sip set debug ip 192.168.0.35
SIP Debugging Enabled for IP: 192.168.0.35
DENG*CLI>
<--- SIP read from 192.168.0.35:5060 --->
REGISTER sip:192.168.0.17 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK53ede5c3795d043a
From: "NH" <sip:[email protected]>;tag=fe3d5e06dd06deb8
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060;transport=udp>
Supported: path
Call-ID: [email protected]
CSeq: 20001 REGISTER
Expires: 3600
User-Agent: Grandstream GXP2000 1.1.5.15
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.0.35 : 5060 (no NAT)
<--- Transmitting (no NAT) to 192.168.0.35:5060 --->
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.35:5060;branch=z9hG4bK53ede5c3795d043a;received=192.168.0.35
From: "NH" <sip:[email protected]>;tag=fe3d5e06dd06deb8
To: <sip:[email protected]>;tag=as5ce077d8
Call-ID: [email protected]
CSeq: 20001 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: REGISTER)
Really destroying SIP dialog '[email protected]' Method: REGISTER
Any ideas what I am doing wrong here?
Nick