If all of your sites are VPN'd together and subnet-to-subnet communication between sites is transparent over the VPN connection then there should be no need to be messing with the SIP packets in the firewall.

That document is the framework for a fault resistant sipX setup. While the setup you have outlines will work just fine, if one of your site links goes down then your phones will too. With a redundant setup like the one outlined in the DNS document if a site link goes down users will still be able to perform basic calling scenarios and dial 911, which in most cases is very important. My suggestion is to buy an FXO gateway for each location, set up the emergency dial rule with location based settings for each FXO gateway and see if you can get the local telco to install a 911 only phone line for cheap. In my case I have to pay for a full local line every month per location (upwards of $45 a month) but it has proven worth it's salt to have a secondary server and FXO gateway when there was actually a situation where the network was down to the site and there was a medical emergency.

You can still keep the main installation in the datacenter, but I'd recommend putting small inexpensive secondary servers in place for redundancy. Out of curiosity how many extensions are you running at each site, and how much bandwidth do you have between your sites?

Nathan Nieblas wrote:
Based on that DNS document, my assumption is that I should have a local
sipX box at each location rather than a cluster residing in the
datacenter? Polycom's are on 3.2.2, I will downgrade them to 3.1.3c and
see what happens. All phones are showing up registered. What I mean by
SIP inspection is that the firewall is essentially handling the SIP
protocol for translation sanity (since it's being NAT'd) and security.

-----Original Message-----
From: Josh Patten [mailto:[email protected]] 
Sent: Monday, January 04, 2010 8:21 PM
To: Nathan Nieblas
Cc: [email protected]
Subject: Re: [sipx-users] multi site deployment

http://wiki.sipfoundry.org/display/xecsuserV4r0/Setting+up+BIND+with+loc
ation+based+views+for+sipX 
this may be helpful for you to implement as it creates a much more 
redundant setup

As far as the other issues you are experiencing, make sure your Polycom 
phones are on a firmware revision no later than 3.1.3c. 3.2 and up has 
known issues with sipX and should be avoided until sipX version 4.2 is 
released. Also, are all of your phones showing up in the registration
table?

What do you mean by "SIP injection"? sipX and sipXbridge generally work 
best when the SIP messaging isn't messed with by a third party product.

Nathan Nieblas wrote:
  
I come from a Cisco Call Manager background and I am trying to apply
    
the
  
same centralized concept/design for a sipX deployment...  I'm just
    
gonna
  
throw what I think would be all relevant facts out there and hopefully
one of you guys can point me in the right direction. Thanks in
    
advance.
  
Scenario: 
1 Head office, 2 branch offices, 1 Datacenter
All locations are connected by VPN tunnel.
Mixture of Polycom and Cisco phones
PBX located in datacenter, virtualized on ESXi 4.x

We are randomly having issues with 1-way audio after putting a call on
hold and resuming it or being picked up from call park. The issue
appears to be related with calls that originate from the Auto
    
Attendant.
  
Another issue appears to be that Polycom phones cannot dial extensions
on Cisco phones and the call ends up in voicemail but works the other
way around. I have all phones pulling their configuration from the
    
TFTP
  
server on the sipX box so I'm assuming there shouldn't be any URI/SIP
preferences missing that would keep the phones from talking the same
language.

This is my digitmap on the Polycom's, our extension ranges are
    
100-113,
  
200-205, 400, 700-705 and 1200-1203:


    
[2-9]11|0T|10x|11x|[2-7]xx|120x|*xxxx|011xxx.T|9011xxx.T|1[2-9]xxxxxxxxx
  
|91[2-9]xxxxxxxxx|9[2-9]xxxxxx|[8]xxx

The PBX assigned a private address and is NAT'd behind a Cisco ASA
that's handling all the SIP inspection, Bandwidth.com is our trunk
provider. I have calls delivering over port 5080 to a local sipXbridge
and I have SIP inspection also being done over that port with the ASA.
    
I
  
have nothing special configured for inspection.

What could I be doing wrong here? 

Nathan Nieblas
SACA Technologies, Inc.

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