How to connect extension

FreePBX Installation / Upgrade
1

Hi,

I’m a new FreePBX/Asterisk’s user. I’m just installed AsteriskNow.

Next, I created an extension in FreePBX with the minimum :
-Extension: 100
-DisplayName: 100
-secret: 100abc

And now I’m trying to connect with diffrent equipments (Linksys PAP2T, Twinkle and a Siemens IP Phone)
But all the equipment refuse to connect:

Linksys PAP2T say: Can’t connect to the server

Here the Twinkle’s log:

+++ 5-9-2010 20:27:12.549979 INFO SIP ::send_sip_udp
Send to: udp:192.168.2.200:5060
REGISTER sip:192.168.2.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.112;rport;branch=z9hG4bKrllevesu
Max-Forwards: 70
To: sip:[email protected]
From: sip:[email protected];tag=kxhpq
Call-ID: [email protected]
CSeq: 178 REGISTER
Contact: sip:[email protected];expires=3600
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0

+++ 5-9-2010 20:27:12.555535 INFO NORMAL ::listen_udp
Received ICMP from: 192.168.2.200
ICMP type: 3
ICMP code: 3
Destination of packet causing ICMP: 192.168.2.200:5060
Socket error: 111 Connexion refus…e

+++ 5-9-2010 20:27:12.555630 INFO NORMAL t_tc_non_invite::process_icmp
ICMP error received.

+++ 5-9-2010 20:27:12.555724 INFO NORMAL t_tc_non_invite::process_failure
Transaction failed.

Send internal:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.112;rport;branch=z9hG4bKrllevesu
To: sip:[email protected];tag=jwsis
From: sip:[email protected];tag=kxhpq
Call-ID: [email protected]
CSeq: 178 REGISTER
Server: Twinkle/1.4.2
Content-Length: 0

+++ 5-9-2010 20:27:12.561895 INFO NORMAL t_phone_user::handle_response_out_of_dialog
Failover to next destination.

+++ 5-9-2010 20:27:12.562272 INFO SIP ::send_sip_udp
Send to: udp:192.168.2.200:5060
REGISTER sip:192.168.2.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.112;rport;branch=z9hG4bKqcqimmpt
Max-Forwards: 70
To: sip:[email protected]
From: sip:[email protected];tag=kxhpq
Call-ID: [email protected]
CSeq: 179 REGISTER
Contact: sip:[email protected];expires=3600
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE
User-Agent: Twinkle/1.4.2
Content-Length: 0

+++ 5-9-2010 20:27:12.565132 INFO NORMAL ::listen_udp
Received ICMP from: 192.168.2.200
ICMP type: 3
ICMP code: 3
Destination of packet causing ICMP: 192.168.2.200:5060
Socket error: 111 Connexion refus…e

+++ 5-9-2010 20:27:12.565178 INFO NORMAL t_tc_non_invite::process_icmp
ICMP error received.

+++ 5-9-2010 20:27:12.565268 INFO NORMAL t_tc_non_invite::process_failure
Transaction failed.

Send internal:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.2.112;rport;branch=z9hG4bKqcqimmpt
To: sip:[email protected];tag=tdtay
From: sip:[email protected];tag=kxhpq
Call-ID: [email protected]
CSeq: 179 REGISTER
Server: Twinkle/1.4.2
Content-Length: 0

I tried to install AsteriskNow on different computers and virtual machines but the problem is the same

Thank’s for help me
Yenyen

PS: Sorry for my english :wink:

2

Hi,

From the trace, the 192.168.2.200 (the asterisk box?) is denying SIP connections on port 5060.

This maybe says that the sip protocoll is not running in asterisk or you have a firewall problem in asterisk box.

You can check the SIP by entering “core show channeltypes” in an asterisk shell, it must be listed in the output. “Sip show settings” will also tell if the correct port 5060 is up too.

In asteriskNow, the you can configure the firewall by entering “setup” in the shell, and disable the firewall temporaly, until you check the config.

I am new to freepbx too, and maybe i cant be much help, but I think you should check these first.

Kostas

3