SIP Registration Error

[udy786]

Hi,

I am trying register a extension from eyebeam softphone from different location on static IP. I have configured eyebeam correctly with user name, password and domain but getting "Registration Error, 408 - Request TimeOut". From same PC I am getting ping for server. Firewall is off.

When I am trying to register that I can see SIP request on CLI.

Below are logs:-

[Apr 30 03:01:59] NOTICE[3266]: chan_sip.c:17601 sip_poke_noanswer: Peer '301' is now UNREACHABLE! Last qualify: 0
[Apr 30 03:02:00] -- Registered SIP '301' at 192.168.1.160 port 11073


SIP SHOW PEERS:-

301/301 192.168.1.160 D 11073 UNREACHABLE

From same PC for different server I am able to register. So as per me there is no any issue with PC or setting of PC because if its working for other server but only working for a server. Please correct me if I am wrong.

Problem is, if I am trying to register from remote location then server have to get static of that location but here I am getting local IP of PC i.e 192.168.1.160.

Locally I am able to register on server using eyebeam.

Server have two different IP. So I can server locally with local IP and from out side the office I can access using static IP.


Server:-
CentOS:- 5.6
Vicidial:-2.4
Asterisk:- 1.4
Installed from Goautodial CD.
Goauto Version 2.1

[williamconley]

I think we need some clarification on the statement "firewall is off".

The firewall in the workstation itself (ie: windows firewall)?

The firewall in the Vicidial server itself (ie: iptables)?

The firewall at the workstation's location (ie: router for the workstations/office)?

The firewall at the Vicidial location (ie: router for the Vicidial server ... if there is one)?

Does your Vicidial server have a Public IP address directly or is it attached to a Router that has the IP?

[udy786]

1. The firewall in the workstation itself (ie: windows firewall)?
Yes Firewall is off

2. The firewall in the Vicidial server itself (ie: iptables)?
Yes iptables is stopped.

3. The firewall at the workstation's location (ie: router for the workstations/office)?
That PC is directly connected with router and on router everything is allowed.

4. The firewall at the Vicidial location (ie: router for the Vicidial server ... if there is one)?
Server have static IP and its opened for all without any restriction and all ports are open.

Yes on server have a public IP (Static IP) directly.

Today this problem is solved. I have callled to Digium support. We have G729 licenses from Digium so they have gave me a advice on this problem. According to them if my PC is inside the router then need to set nat value is yes on server. Its not required or compulsory but some it may be create problem. I have changed nat value on server and problem solved.

Its surprise for me. Its was running before some time. No any changes was on server, router and on PC at both end. I have changed nat value and its started. Some time server behaving unacceptable. Really hard to understand.

[williamconley]

We've seen that happen, but usually it involves a carrier "suddenly" changing their server configuration.

Yes the NAT setting should be YES when you are NATting (which is defined as "there is a router between my phone and my asterisk server"). Additionally, you may also need an "externip" setting in many cases (if the router is at the asterisk server's location).

Do note that the reason for this is that there IS a firewall blocking packets in your router. While you may not think you have it turned on, it is impossible for all packets arriving at the router to pass through to your phone unless your phone is the DMZ for the router (and then only ONE phone would get ALL the packets, not entirely viable). So the router must have a method to decide which packets go to your phone and which packets go to other phones or destinations. This makes it operate as a firewall, and it cannot be "off". In the end, the rules in this firewall cause the conflict.

When your phone sends packets to the asterisk server, it includes its own IP in the packets. Asterisk then sends return VOIP packets to that IP. But that IP is a LOCAL address ... and that local address does not exist at the Asterisk server so the packets get discarded by the asterisk servers local network. This results in "UNREACHABLE". Solution: NAT=Y. Now the phone will send the PUBLIC IP (or the asterisk server will ignore the local IP, depending on the circumstances) and the packets will instead be sent to an RPort on the outside of the router on the proper IP. The router will receive those packets on that special port, which is assigned to your phone ... and now the round trip works and you get registration (and sound, if all goes well!). But without that NAT=Y setting, the router complicates the SIP packet transfer to the point of failure.