Carrier Unreachable NAT Issue

[DefLeppard]

I have a Goautodial 2.0 installation which was working fine for a few weeks and then one day out of the blue, the carrier won't register.

This server is behind NAT and DMZ is enabled on the router to point all traffic to this server. I am able to ping the provider but Asterisk won't register. I tried everything in my reach to fix this but couldn't, I checked the router and firewall settings, added externip, enabled nat=yes, but nothing works.

The carrier keeps showing Request Sent in sip show registry and UNREACHABLE in sip show peers.

When I enable sip debug, it keeps showing a bunch of Reliably Transmitting #1 (NAT) messages.

I have had this issue before and was unable to fix it. There has to be a way to fix it.

Please help me. Thanks in advance.

[boybawang]

can you try to ping the carriers ip from your server? is the carrier ip a domain name or just an ip address? make sure 5060 and 5061 udp/tcp is forwarded to your asterisk server

[DefLeppard]

I can ping the carriers just fine. The carrier is a domain name, I also tried using the IP address but it won't work. This happens with all carriers not just one.

When I reboot the server, the carrier shows Registered for a minute or so and then again it becomes Unreachable.

All ports are forwarded to the server, the server in in DMZ of the router.

Also, the agents can register on their soft phones in the same network. But Asterisk won't register. This is the strangest thing. I have tried with Asterisk 1.4 and 1.6.

Any ideas?

[boybawang]

try using the ip address instead of the domain, then put a qualify=1000 , try not using DMZ but just port forwarding first make sure ports 5061 5060 10000 - 20000 are forwarded to the asterisk server

[williamconley]

or try qualify=no

what does

Asterisk won't register

mean? have you been watching the registry and cli output to see what is happening?

also: Vicidial does not work with asterisk 1.6, but if you tried that as a "test only", you appear to have an issue with your router. have you tried another router?

also check your /var/log folder for errors and failures. you may be experiencing an attack of some sort that is overloading your routers nat table (dmz is not a great idea). consider using iptables to lock down your server and see if you can get your IP address changed by your ISP after you have "lockdown" so you can't be detected any more when on your new ip

[scenarist]

I connected directly from my office(vicidial server) to my provider via optical cable. On my side I have a optical converter and connect my lan cable directly from optical converter to vicidial server in NIC eth0. (this NIC is only for telephony i.e sip trunk, another nic eth1 is for my LAN i.e for my agents)
Configuration of eth0 is

Code: Select all

IP: 217.75.206.97
SUB: 255.255.255.192
G: 217.75.206.65


sip debug show me:

Code: Select all

[Feb 15 16:54:14] WARNING[4697]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x844ce54 (len 498) to 217.75.205.49:5060 returned -2: Network is unreachable
[Feb 15 16:54:18] Really destroying SIP dialog '4067fd5168c653d7120c84dc7356f14a@127.0.0.1' Method: OPTIONS
[Feb 15 16:54:28] WARNING[4697]: acl.c:541 ast_ouraddrfor: Cannot connect
[Feb 15 16:54:28] Reliably Transmitting (NAT) to 217.75.205.49:5060:
OPTIONS sip:217.75.205.49;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3ba094c0;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as660bb335
To: <sip:217.75.205.49;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 21df3e4e5a359b72289fd2c7230f5dde@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 15 Feb 2013 21:54:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[b][Feb 15 16:54:28] WARNING[4697]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x844ce54 (len 498) to 217.75.205.49:5060 returned -2: Network is unreachable
[Feb 15 16:54:32] Really destroying SIP dialog '21df3e4e5a359b72289fd2c7230f5dde@127.0.0.1' Method: OPTIONS[/b]
go*CLI>


sip show peers

Code: Select all

go*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status

8002/8002                  192.168.1.6      D   N      19243    OK (110 ms)
8001/8001                  192.168.1.7      D   N      13266    OK (104 ms)
logosoft                   217.75.205.49        N      5060     UNREACHABLE
21 sip peers [Monitored: 2 online, 19 offline Unmonitored: 0 online, 0 offline]


What could be the problem , please help ??

[scenarist]

Uhhh. I SOLVED my problem. Problem was about nic A and nic B. Nic A have a two ports , eth0 and eth1 and nic B also have a two ports eth2 and eth3.
First time I connected both lan cables in eth0 and eth1 (i.e I connected all in one nic, nic A) and there was a problem!

Correct way is:
I connected one lan cable from router to eth0 - NIC A (it will be uses for my agents), and second cable from optical converter to eth3-NIC B (it will be uses only for sip trunk) and everything is OK now ! Great!