Still unable to fully register

[LiquidCaffeine]

Ok, below is my setup:

3 SIP trunks
1 - Gizmo5
2 - Broadband.com

5 Remote Clients
1 - Granstream 2000
4 - Xlite softphones

PIAF Network
1 - Server is on a 10.0.1.0/255.255.255.0
1 - Static public IP
1 - Cisco 5500 ASA forwarding all the required ports to the PiaF server

Settings (sip_nat.conf)
externalip=xxx.xxx.xxx.xxx <---- real IP used here
localnet=10.0.1.0/255.255.255.0
nat=yes

Problem
The FreePBX status page shows all online, however, the Asterisk Info page shows all unreachable. Incoming calls are successfully answered by the IVR, but cannot reach remote devices. Calls to those extensions go straight to voicemail. The devices can actually successfully log in to hear their voicemail. Heck, even the little red message indicator lamp on the Grandstream flashes whenever there is a voicemail. Remote users cannot place any outbound calls to either other extensions or telephone numbers.

Some of the tings tried so far
I've tried turning qualify= to yes and no for the extensions without any luck (only changes status from UNREACHABLE to UNKNOWN ins Asterisk Info).

The Bandwidth.com trunks are all IP-IP, but the Gizmo5 requires a registration string which I know works (tried on another PiaF box without a hiccup). Trolling the logs shows:

[2009-06-11 09:54:50] NOTICE[2717] chan_sip.c: -- Registration for 'xxxxx@proxy01.sipphone.com' timed out, trying again (Attempt #68)

Any suggestions from anyone. Please feel free to ask for any information I may have omitted due to my encroaching madness from trying to solve this issue

 

[jbh]

I think you need externip=xxx.xxx.xxx.xxx

i.e. not externalip= ...

 

[LiquidCaffeine]

I think you need externip=xxx.xxx.xxx.xxx

i.e. not externalip= ...

Sorry, typing the post without thinking . The sip_nat.conf is actually:

externip=xxx.xxx.xxx.xxx
localnet=10.0.1.0/255.255.255.0
nat=yes

 

[angoyr]

LiquidCaffeine,

Maybe you can try changing the address of the PIAF server to something other than X.X.X.0. Your network address and the server address being the same might be causing your problem.

Robin A.

 

[LiquidCaffeine]

LiquidCaffeine,

Maybe you can try changing the address of the PIAF server to something other than X.X.X.0. Your network address and the server address being the same might be causing your problem.

Robin A.

The server internal IP is actually 10.0.1.58

 

[angoyr]

Did you look at the log of your router to see if there are any clues in there. Some "ALG" routers block outgoing sip packets. Also you may want to try setting up an IAX2 endpoint to see if you're experiencing the same problem. If AIX2 works, the router is the culprit.

Just something else to try.

Robin A.

 

[jbh]

Remote users cannot place any outbound calls to either other extensions or telephone numbers.

It does sound like a NAT issue in so far as your remote phones can make calls to voicemail at least, but cannot receive calls. Odd that it is happening to so many different phones/clients though. Are they all in a variety of different locations or in the same place?

When you say they cannot place outbound calls to other extensions - do you mean extensions on the same network as the PBX as well?

Are you able to plug a couple of the phones in on the LAN that the PBX is on to see if they can each other then?

Odd that remote phones cannot call out on the trunks. Are phones on the LAN with the PBX able to do so?

 

[jbh]

Robin's theory sounds a very good one

I missed his post while writing mine !

 

[LiquidCaffeine]

It does sound like a NAT issue in so far as your remote phones can make calls to voicemail at least, but cannot receive calls. Odd that it is happening to so many different phones/clients though. Are they all in a variety of different locations or in the same place?

When you say they cannot place outbound calls to other extensions - do you mean extensions on the same network as the PBX as well?

Are you able to plug a couple of the phones in on the LAN that the PBX is on to see if they can each other then?

Odd that remote phones cannot call out on the trunks. Are phones on the LAN with the PBX able to do so?

Sorry, I should have clarified a bit. All local LAN units (either X-Lite or Grandstream 2000s) work without a hiccup for extension to extension dialing. Inbound calls to local extensions work as well. Remote users and all trunks show registered in the FreePBX dashboard but unreachable in Asterisk Info, however, inbound calls routed to local extensions ring, but external extensions, if selected, go straight to voicmail. Also, whenever a call is placed outbound whether by a local or remote user, the "All circuits are busy..." message is played by the server.

I agree with Robin in that it may be a firewall issue, so am having the Cisco guru at the office inspect it closer. Am also looking to try to connect via a know working external PiaF server via IAX to see if that connects correctly. Heck i need to do that anyway since it is a remote office and once this server is up and running want to try and use its SIP trunks at the remote office.

Whew. This is only my third install and the only one that is being a beast - thanks Cisco.

 

[jbh]

It does sound like the firewall is allowing incoming SIP packets from a random remote port to its port 5060 (presumably as you have that port forwarded to the PBX) but is not allowing traffic outwards to port 5060 - hence the trunks fail also.

You could also try registering a phone located on the LAN to a remote SIP VoIP service to verify this. If the theory is correct, this should fail also.

 

[LiquidCaffeine]

Did you look at the log of your router to see if there are any clues in there. Some "ALG" routers block outgoing sip packets. Also you may want to try setting up an IAX2 endpoint to see if you're experiencing the same problem. If AIX2 works, the router is the culprit.

Just something else to try.

Robin A.

Robin, you were right on the money. As soon as we disabled Cisco's SIP inspection features and switched to just straight port forwarding, everything fired up without a hitch and has been running great for the last 3 days.

 

[angoyr]

Hi,

We're happy that you got the problem solved. Good luck with the rest of the project.

Robin A.