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Look into a FRee space optical wireless solution. You can push gige speeds in monsoon like conditions through some of these products. The companies to look at are MRV, CANON(they are the newest into the market but have become the largest player. Lightpoint, laserbit, and lightpoint. All of...

pretty vague.. what kind of phones.. what kind of switch.

First if it is working why do you want to do it? Second what platforms?

what type of router do you have at the remote site? You could use the router and not the pix to support the VPN. Also the new softphone is as the 7970 phone running on a PC.. IT works amazing and will solve your VPN problems.

You must be using a traslation pattern to match your call routing information in unity.

Pretty straight forward.. I would look into IOS issue..

VOIP trunk.. Ie h323? Has it been locking up recently? There is an IOS dos attack that came out a few weeks ago targeted towards voice IOS's

Ahh DMS100 has many problems on dt24+.. esp with nortel!!! Did not notice in orig post you used DT24. Change to ESS5.. Set DT24+ to isdn network..Set pbx trunk to ISDN user.

How is the PRI terminated? MGCP or H323? IF MGCP are you binding q.931 to ccm via backhaul support?

Now that I think about it.. region configs are bound to each other.. Once you change one direction it automatically changes the other! The phones must use the same codec... I don't think you have a choice

computerhighguy, Have you ever verified this? Press the I button twice while you are in a call and see if you ever see different codecs on the phones without the use of a transcoder over a trunk. The reason I don't think this works is all the regions really due is supply the H.323 Proxy(cisco...

In my opinion on this. Why have your Fax server integrate with CCM or unity!!! Run tone detection on your Analog gateway(5400 works well for this) and route faxes directly to faxserver and calls directly to softswitch.

computerhigh guy.. Have you verified seperate upstream and downstream codec? I don't that actually works.. What I am sure you can do though is set up a call so that call dialed from one side of the call will settup with a codec differnt then calls originating from the opposite end. But once...

what exactly are you trying to do?

How many analog voice ports are you looking for? All cisco router DSP with enough PVDMS will support g.729. It is actually the default codec for IPV4 peers! You can get down to about 15K per call running RTP header compression. on PTP links with a decent router this actually works quite well...

Go 3750 stackwise across the board. Two GIGE fibre only for the core or your network.. ANd add as many as you want to your closets.. Quite a capable piece of Gear.

Per Katzeyes original post VM is required. I have rented a facility like this and they do usually provid receptionist and VM service.

how much is a "small office" IP office settup with VM?

You need to allow UDP packets greater than 1024 past your router.. You are not allowing UDP responses back through right now. Do a show ip access-list and place here.. But overall for what it appears you are trying to do. YOu want a FW.

Cef does load balencing by default.. But on a per flow basis.. Why are you trying to change this?