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Cheap Site-to-Site VoIP solution

Hi everyone.

I'm looking for a cheap VoIP solution to make calls between 2 sites. These are actually been done using two 2800 ISR routers with VICs, acting as h.323 gateways. The sad story is that they have to go , because they're owned by our MPLS provider and we are changing these to plain T1 links.

I think I can achieve this with a couple of UC500, as they still have H.323, and can act as gateways. I can even copy part of my current config and place it on them. But I'm wondering if there is a cheaper way to achieve this.

I currently have two old 1700 series routers with VIC cards, previously used for Frame Relay links, actually owned by the same provider but they never asked us to return them . I think they are going to take our 2800s away and maybe those 1700s too, and place 1800s instead.

I'm looking for the most cost effective solution to make calls between the 2 sites across a VPN tunnel (which will be in place using another equipment). My choices are:

1. Use the old 1700s as H.323 gateways (if they don't take them away :D).

2. Buy two VICs and place them into the 1800s.

3. Buy two (insert-best-cost-effective-solution-here) routers to handle VoIP across the VPN tunnel.

For instance, my current dial plans (voip) look like this:

dial-peer voice 400 voip

destination-pattern 4..

voice-class h323 1

session target

incoming called-number .

codec g729br8 bytes 50

fax rate 9600

ip qos dscp cs3 media

ip qos dscp cs3 signaling


TDM Extensions are: 1xx, 4xx and 88x are in one (local) site; and 2xx, 3xx and 64x are in the other (remote) site, every phone is still analog.

Can I achive this with uc300, or something cheaper than the uc500? I don't want to buy discontinued (marked as end-of-sale) devices like SPA9000. Can I buy VICs to the new 1800 routers and place my current configs to them? Which is the best choice?

Please, help me make my choice.


Oscar Mascareñas

David Trad
Rising star

Hi Oscar,

I am going to have to say NO the UC-300 is not equipped just yet for this to happen, In time I hope they allow for SITE-2-SITE calling with SIP trunks as this little system is more than capable of doing it. Right now it is not an out-of-the-box feature.



Cheers, David Trad. **When you rate a persons post, you are indicating a thank you or that it helped, but at the same time you are also helping to maintain the community spirit - You don't have to rate posts and you wont be looked down upon :) *