Boss is now 'sold' on using the emergency POTS line as a backup. I just 
needed to take a different angle :)
Thanks.

On 2/1/2010 7:23 PM, [email protected] wrote:
> Thanks. Tools for my toolbox is what I need right now.
> I guess I'm looking for that perfect mix of decentralized vs 
> centralized. I need to have as little as possible running at the 
> remote facility. This goes way beyond just telephony. With that being 
> said I'm going to take the points presented by you guys back to my 
> boss. Thanks for many of those points Tony. Given the price of a small 
> FXO device, I think I can make the sell for it. I can probably sell it 
> on the point of good PR (in case the circuit goes down) over safety. I 
> know it shouldn't be that way, but it is for me.
> So, with trying to keep as much central as possible, if I configure 
> nothing but sipxbridge at the remote sight, can a I keep calls to and 
> from the PSTN from coming back through my central server? I would have 
> the phones registering with and using voicemail on the central server. 
> I would specify the appropriate sipxbridge instance in a dial plan for 
> each group of users. Does that work from a network flow point of view? 
> If not, skip the next paragraph.
> If the above works (and I've sold my boss on POTS backup), I need to 
> configure the handsets (Polycom 450s) to use an FXO gateway if the 
> MPLS connection/sipxbridge gateway is down. I'm slightly partial to 
> the Audiocodes MP series. I got one working for inbound backup in 
> about 20 minutes, so that is a testament to Sipx's ability to manage 
> it! Is it possible to configure the Polycoms to use an Audiocodes FXO 
> directly with their primary line for outbound calls if they can't get 
> out through sipxbridge? If not, is their a minimal sipx setup I can 
> use as a secondary sipx server that the handsets could use if the 
> primary isn't available?
>
> Thanks as always,
> Matthew
>
> On 2/1/2010 5:37 PM, Eric Varsanyi wrote:
>> In the polycoms (and I presume pretty much any other multi-line 
>> phone) you can register external lines that register direct from the 
>> handset to a voip provider. SipXecs manages the phone configuration 
>> in this case but otherwise doesn't get involved. The normal sipXecs 
>> lines continue to work normally.
>>
>> This wouldn't  be seamless but it might be useful in your situation 
>> as the talk path would go direct from the branch office to Verizon. I 
>> use this sort of setup on the 650's as a backup in case the sipXecs 
>> server goes down and I want to dial 911 via a local Patton bridge. 
>> Unless verizon supports it you wouldn't get forward on no answer to 
>> voice mail, likely that would kill the idea right there. Verizon may 
>> also not support multiple registrations on a given DID number so each 
>> user would need a separate external phone #.
>>
>> Likely its not appealing at all, just a possible tool in the toolbox.
>>
>> -Eric
>>
>>
>

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