Hi Here is my setting.
PBX Firmware:
1.0.0.0
PBX Service Pack:
1.88.210.57-1
ISP : Time warner cable
Freepbx has a local IP address is 192.168.0.8
I have forwarded UDP 5060 and 5061 to the 192.168.0.8
I have forwarded UDP 10000 to 20000 to the 192.168.0.8
All I have done so far is created a couple of extensions. I cant seem to get my linksys Pap2 ATA register successfully to it. I have enabled NAT in all the extensions.
I have DynDns setup so that I have a public DNS available and Im able to ping to that.
In the proxy field I put my public DNS.
I have a syslog server running in the same network as the freepbx. and I have forwarded port 514 on my router for syslog to that PC.
I can see that request from the remote extension coming to the pap2 and the response from the pap2 via the syslog console.
Here is what I get from the syslog server console
19:09:55 117.193.67.226 Feb 7 06:40:13 0016B65E35A8 SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 192.168.2.2:5062;branch=z9hG4bK-4de07e9e;received=117.193.67.226
From: MyPBX sip:[email protected];tag=e521e88c8c6391ddo0
To: MyPBX sip:[email protected]
Call-ID: [email protected]
CSeq: 160 REGISTER
Server: YATE/3.3.2
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
Content-Length: 0
19:09:55 117.193.67.226 Feb 7 06:40:13 0016B65E35A8 SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 192.168.2.2:5062;branch=z9hG4bK-9140794;received=117.193.67.226
From: MyPBX sip:[email protected];tag=e521e88c8c6391ddo0
To: MyPBX sip:[email protected]
Call-ID: [email protected]
CSeq: 34206 REGISTER
Server: YATE/3.3.2
Allow: ACK, INVITE, BYE, CANCEL, OPTIONS, INFO
Content-Length: 0
19:09:55 117.193.67.226 Feb 7 06:40:13 0016B65E35A8 REGISTER sip:mypbx.getmyip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.2:5062;branch=z9hG4bK-4de07e9e
From: MyPBX sip:[email protected];tag=e521e88c8c6391ddo0
To: MyPBX sip:[email protected]
Call-ID: [email protected]
CSeq: 160 REGISTER
Max-Forwards: 70
Event: keep-alive
User-Agent: Linksys/PAP2-3.1.12(LS)
Content-Length: 0
19:09:55 117.193.67.226 Feb 7 06:40:12 0016B65E35A8 REGISTER sip:mypbx.getmyip.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.2:5062;branch=z9hG4bK-9140794
From: MyPBX sip:[email protected];tag=e521e88c8c6391ddo0
To: MyPBX sip:[email protected]
Call-ID: [email protected]
CSeq: 34206 REGISTER
Max-Forwards: 70
Contact: MyPBX sip:[email protected]:5062;expires=3600
User-Agent: Linksys/PAP2-3.1.12(LS)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
I have enabled SIP debug on from the CLI.
I dont see any activity on the asterisk CLI.
In addition to the port forwarding in my router, do I need to do anything in the freepbx server?
I see that fail2ban is running, is that something I mess with.
I have been using asterisk for couple of years now and I always put my server and DMZ and have not dealt with these issues. My Timewarner cable has a DMZ option but does not allocate a PUBLIC ip to my freepbx server. So Im trying to get it working with just the port forwarding.
This is driving me crazy…
Any help is really appreciated…