sundance-communications.com/forum Forums TDM Business Telephone Systems Toshiba Telephones & Systems LIPU and Toshiba SoftIPT

Hi, we have a LIPU and we have a pc outside our network that connects into the network using Cisco VPN software over the internet to a Cisco Pix firewall. We have the Toshiba SoftIPT software installed and it can dial and recieve calls on its assigned extension. However, no audio is heard through the handset and the person on the other end does not hear anything. We have a televoip handset connected straight to the pc, which basically had the red and green plugs that go straight into the soundcard. I am able to open windows sound recorder on the pc and record my voice through the handset and it plays back through the handset fine, there is just no audio sent or recieved through the handset when a call is made through the SoftIPT software. Is anyone familiar with this at all or know if we need to open any ports or do anything special to the Pix firewall or anything? we are pretty baffled, any information would be great.

I think I recall this being fixed by a firmware upgrade for the LIPU.

I will check in more tomorrow, but until then, do you know your system software version, as well as your LIPU firmware version?

make sure the Codex is G.711, not G.729A

Thank you both for the responses, however, we can't figure out where to go to find the information you guys are looking for. We logged into Emanager and then connected to our system in Emanager and it has the Emanager version and the CIX/CTX version which is: AR4.10MN023.00
Poking around in the menus on Emanager we were unable to find the information you were looking for, where should we have gone?
Thanks in advance.

For Firmware, go to Advanced, IP-Telephony, IPU/IPH Config (Or may say IPU Config). Select the card from the list and the screen should populate.

To check the codex, go to Advanced, IP-Telephone, IPT Data (Program 250). Select the extension from the list on the right. The screen should populate. On 08 Codex should be G.711 for softphones. IPTs can be G.729A. Secondary codex (14) should be same or none.

If you have problems with audio from on IP phone to another, you can try enabling #16, No Peer to peer. Usually you only need this if the phone is outside the LAN. It should not matter if you are in through VPN.

Disabling peer-to-peer might do the trick as I remembered having to disable it on a VPN once.

Good to know. Have not done any IPT's thru a VPN yet. I have usually just put the IPU on the public side of the router.

Other things that sometimes interfere with the audio is the QOS settings, diffserv and 802.11P and 802.11Q.

Same here... I haven't installed a single IPT through a VPN yet, every single one has been public IP, ahead of the firewall.

thx phonemeister.

Can you explain your comment of disabling peer-to-peer?
I am having the same issue as the original poster of this thread.

Boy, this is a pretty old post. February of 2008?

As mentioned, in the Toshiba phone system programing (not settings in the Soft Phone), you would ENABLE a setting that says IPT no peer to peer. This forces all audio through the IP card instead of trying to communicate directly between IP phones or other Toshiba IP devices. There was a very recent post talking about this setting: https://sundance-communications.com...eceiver_can_hear_caller_but_n#Post541259


Usually you can leave it default if your problem is not talking to other IP phones. Total audio failure for all calls is generally due to other network issues, but it is something you can try.

There are many other things that could be causing audio problems through a VPN tunnel, so I wouldn't get your hopes up that this will be the magic bullet. I spent about 3 hours with Soniwall support before I got phones working across customer's VPN. Typical problem is ports being blocked, even when the IT person swears that all the phones have total access across the VPN.

Last time I tried to use a Cisco VPN client (many years ago), I found it impossible to use it for any VoIP applications. One problem was that there was no route from the site's network back to the client PC. It was more or less a one way street. VoIP is a 2 way application. Hopefully it works better today.