Morning all.

In the snow we all have our LAN-LAN and remote worker connections from our home based 2820s to the office 3300.

The data side is fine. We have a Panasonic KX-TDE phone system with three VoIP handsets located at the ends of three LAN-LAN connections. These work fine to a point. we can do everything that we would expect to be able to do except talk external VoIP extension to external VoIP extension.

I can ring a home extension and the connection is made, but there is no audio in either direction. Calling from external VoIP to an internal extension, or dialling out or receiving a call on a CO line are all fine.

We are assuming that we need to set up some extra routing in the 3300?

Any thoughts please?

Thanks in advance - Terry

Happy New Year!

If the same config phone was in the office does it work?

Can the PBX be seen from the remote VPN's for the protocols used?

If your phones are using SIP & RTP over the VPN's then checking you can see this traffic at the PBX?

If the 2820's are sitting on DSL links then you might also check that:

- they have enough upstream bandwidth.

- that you don't have high ATM error rates on the circuits.

- That your using an efficient codec (e.g. G.729)

- That you have traffic profiling prioritizing the time critical VoIP traffic over the non-VoIP traffic.

Alan

Thanks for the reply Alan.

Yes, the IP phones work fine in the office, but I can't recall if we actually have tried IP phone to Ip phone in the office (as opposed to digital/analogue handset to IP handset). What we have tried is my home IP handset talking to one of the IP handsets when it is in the office (where the phone switch is located).

Your second question is an interesting one; the switch can definitely be seen from the IP handsets for everything (lamp/status keys, call transfer, access to the phone switch directories etc) and the IP handsets can receive calls from the switch so long as the other party isn't an external IP handset. These are proprietory Panasonic VoIP handsets, not their SIP variety.

Re the SIP and RTP, we are starting to think that the switch hands off the conversation once connection has been made. I think that we will need to take a couple back into the office and test them internally. I expect that they will work and the problem will be that the switch hands off the call once connected (this frees up the DSP on the switch) but the call effectively becomes orphaned and the phones don't know how to talk to each other.

Bandwidth isn't a problem and re the ATM rates we may have to check the combined rates between the two VPNs.

We have QoS turned on and that seems to work fine, we have checked it with the other calls and all looks good.

Thanks again for the pointers.

Kind regards - Terry

Update:

We have now tested in the office and the IP extensions work 100% fine.

It is just when they are outside of the firewall that the audio doesn't seem to know how to route.

We are in contact with Panasonic support as well, but no initial brain wace from them yest.

i have the exact same problem, i have a panasonic KX TDE100, 15+ IP phones in one location and a few off site connected via ipsec lan to lan, all work fine, problem i have is that now we are going via a external sip provider all we get is the line no audio, if we go via our old isdn 30 lines no problem at all. The only light i can shed on it from panasonic tech support is apparently the nat is just displaying the internal ip address and not the external ip address and cannot root back the sip service. hope this contributes to some form of solution as its getting a pain now

Are you using the Panasonic SIP (?) or the IP handsets?

Our conversation with PUK support indicates that we are looking for spoke to spoke (as well as spoke to hub) support.

We have had problems with this and Drayteks before (they didn't have the feature supported) and we had to use another approach with other kit. This wasn't for phone support but involved connecting London and Bahrain to New York and then Bahrain being able to see parts of London's network as well.

Does anyone know if the current 2820s (and the 3300) can either support spoke to spoke or even "partial spoke" where the you can address a specific IP address on another remote site.

Thanks