No Dial Tone on 9600 phones That are VPNed 3

[mjoetech]

Hello,

I set up some 9650 phones in the phone room at our office. They were working fine.

I then brought these to our other office which has a VPN tunnel to our main office. I reprogrammed these with the new IP's and they booted up just fine. However, there is no dialtone when i press a line appearance now. I can still call and everything works great except the fact the is no audible dial tone.

Does anyone have a suggestion oof what can cause this?

Regards,

Joe

 

[Mitch672]

absolutley. the "UDP" packets are not being routed properly to the subnet being used for your voice equipment. check your IP routes/ip routing, you will find there is not 2 way communication between the IP segment the VPN phones are being assigned and your phone LAN subnet.

No UDP packets = No Audio

Mitch


Mitch

AVAYA Certified Expert

 

[mjoetech]

Sorry.. These are installed on a g350/8300 CM4.0.

 

[mjoetech]

Thanks very much for this fast and accurate reply!

Joe

 

[Mitch672]

switch doesn't matter, check the IP routing. easisest way is to install "WireShark" on both ends, and watch the particular IP address's involved, I am sure you will see packets/data not making it in both directions.

Also, you might want to upgrade that to CM 4.0.4/Cm 4.0.5, CM 4.0 is pretty well done at this point.



Mitch

AVAYA Certified Expert

 

[mjoetech]

Thanks Mitch. Your talking just a bit over my head on the upgrade procedure, but i'm a sponge to learn this more. Is the an avaya doc that you can point me to for this?

Thanks,

Joe

 

[Mitch672]

You can download the CM 4.0.4 or CM 4.0.5 ISO, but I suggest you enlist the assistance of a locla Business Partner, if you have never done an upgrade. You can read about it in the AVAYA docs as well, but there is no substitute for actual experience

Here are some links:

CM 4.0.5 ISO image:

http://support.avaya.com/css/appmanager/public/support/Downloads/P0001/4.0.x/C201013121359281026

You burn that to a CD, and use that to load an "installable" image to your server(s), then run through the Server Maint screens to install it, yo uneed to of course backup your translations, and your audix, etc. again, read the documentation, and perhaps get a BP to assist for your first upgrade. It isn't that hard, but there are a few tricks and things to remember, like to "make upgrade permananet", and also your switch will be down during this proceedure, so you need to do it on a night/weekend when no one needs the phone system.

Mitch


Mitch

AVAYA Certified Expert

 

[mjoetech]

Well,
I thought it was a fix, but there is still no dial tone on the remote 9650. Coincidentally, I have a definity in the same phone room and a 4612 in the same remote. These are on the same VPN, and the 4612 works well and has dial tone.

I tried comparing setting on the two systems. The only thing I cant change is the Min UDP. The Definity is set to 1716 and I cant go lower than 2048 on the g350.

I graciously accept any recommendations,

Joe

 

[dufus2506]

Have you done a "list trace" to see what happens when the station is used?

Perhaps you have certain UDP ports blocked in your VPN, or you have no access to one of the system boards. (Like a Medpro or something.)

Start with the simple things first.

You might also make sure you have version 3.1 installed for the 9600. It's the latest release, and I believe has changes associated with VPN.

Carpe dialem! (Seize the line!)

 

[mjoetech]

Thanks Dufus2506:

Below is my trace. I picked up the handset on 3607 and called 3600. When I picked up the handset on 3607, I had no dial tone, yet completed the call to 3600 successfully.

Thanks.
LIST TRACE

time data

08:53:39 active station 3607 cid 0x2be
08:53:39 G711MU ssff ps:20 rn:1/1 192.168.26.92:43278 192.168.5.70:2074
08:53:39 xoip: fax:Relay modemff tty:US 192.168.5.70:2074 uid:0x95d
08:53:43 dial 3600
08:53:43 ring station 3600 cid 0x2be
08:53:43 G711MU ssff ps:20 rn:1/1 192.168.5.63:2068 192.168.5.70:2076
08:53:43 xoip: fax:Relay modemff tty:US 192.168.5.70:2076 uid:0x3
08:53:47 active station 3600 cid 0x2be
08:53:47 G711MU ssff ps:20 rn:1/1 192.168.26.92:43278 192.168.5.63:2068
08:53:47 G711MU ssff ps:20 rn:1/1 192.168.5.63:2068 192.168.26.92:43278
08:53:56 idle station 3607 cid 0x2be
08:54:10 rcv KARRQ endpt 192.168.26.92:49300 switch 192.168.5.71:1719 ext 3607
08:54:10 snd KARCF endpt 192.168.26.92:49300 switch 192.168.5.71:1719 ext 3607

 

[dufus2506]

Ok... What I see is that the PBX is trying to connect to your phone's IP on port 43278. The PBX is sending from it's own port 2074.

Is there any reason that port 43278 would be blocked through your VPN?

You should also check the medpro board and Clan boards to make sure they are configured to use a proper network gateway that can route packets to the VPN.

Carpe dialem! (Seize the line!)

 

[mjoetech]

I believe the ports are open.

My question is how do I check for the medpro and Clan on a g350 in order to use a proper network gateway that can route packets to the VPN?

Thanks!

 

[Mitch672]

the answer is a PC with ping, but that will only test ICMP protocol, not UDP. See if you can ping the IPs you need to be able to reach, that will at least test ICMP.

You might also want to consider a VPN client for the phones, 9600 R3.1/3.101a has a VPN client built into each phone, you won't need a site-to-site for it to work, each phone makes its own connection. There is also VPN firmware availble for the older 4600 IP phones as well.

Again, you most likely have a UDP routing issue.

MItch




Mitch

AVAYA Certified Expert

 

[dufus2506]

To check the IP configuration of your various boards, type the command:

change ip-interface <location>

Check the network gateway assigned. Make sure it's correct.

Carpe dialem! (Seize the line!)

 

[boldweb]

I traced the port activity from the VPN router itself, and it showed the UDP traffic going through. Any other ideas why this is not getting audio for dial tone?

PS. mjoetech and I are working out of the same office on same project.

 

[dufus2506]

Are these stations SIP or H.323?

Is there any NAT happening in the network?

Part of a SIP packet contains the IP address of the originating node. Even if a router or firewall re-addressed the packet, the payload IP will still be wrong.

This can cause the PBX to send the information to the wrong IP. (The local IP of the phone, instead of the NAT address the PBX would see from it's side of the network.)


Carpe dialem! (Seize the line!)

 

[boldweb]

The stations are H.323, does this make a difference?

 

[Gibsonic]

i have been trying to figure out the reason for not getting dial-tone on an initial line appearance selection for a while and today i figured it out.

i did a trace on the station and the reason became very apparent.

On first press, the line appearance tried to originate from a network region that was configured for G.729a as its codec.

On second press, the line appearance tried to originate from a network region that was configured for G.711 as its codec.

I had not created a specific network map for the subnets being used by the VPN phones.

I set the VPN phone subnets to NR 1 which has G.711 as the codec and now even without rebooting the phones they work every time with beautiful clean dial tone at each press.

 

[mjoetech]

I did this as you stated and still no dial tone when I go off hook on sta 3698. Everything else related to the call is fine. Just no dial tone on the remote phone. I did a list trace and posted below. Any suggestions? Thanks!

time data

10:19:59 active station 3698 cid 0x19b
10:19:59 G711MU ssff ps:20 rn:2/1 192.168.26.92:2328 192.168.5.70:3310
10:19:59 xoip: fax:Relay modemff tty:US 192.168.5.70:3310 uid:0x95c
10:20:00 dial 777 route:UDP|AAR
10:20:00 term trunk-group 1 cid 0x19b
10:20:00 dial 777 route:UDP|AAR
10:20:00 route-pattern 20 preference 1 cid 0x19b
10:20:00 seize trunk-group 1 member 3 cid 0x19b
10:20:00 Calling Number & Name NO-CPNumber NO-CPName
10:20:00 Setup digits 777
10:20:00 Calling Number & Name NO-CPNumber Mike G350 Sta
10:20:00 Proceed trunk-group 1 member 3 cid 0x19b
10:20:01 Alert trunk-group 1 member 3 cid 0x19b
10:20:01 G711MU ssff ps:20 rn:1/1 192.168.5.21:2204 192.168.5.70:3312
10:20:01 xoip: fax:Relay modemff tty:US 192.168.5.70:3312 uid:0x50003

 

[Avayapbxtech]

I have a G350 with an 8300 installed and we are not receiving a dial tone although we are able to call out. We don't have any Network regions setup because it is just a standalone system. We were receving dial tone previously, but recently we noticed the issue. Any assistance would be appreciated.