Struggling with remote phones

[grnerd]

I am struggling with one sided calls with remote phones. In this setup, there are 3 different physical sites. The sites are all connected via site to site vpn tunnels. The remote sites have multiple Allworx phones. I have assigned seperate IP addresses on each remote phone, and assigned SIP ports and RTP ranges via a unique handset preference group for each phone. I have opened ports on the firewalls to push this SIP and RTP traffic to each individual phone. I have 2088 and 8081 forwarding to just one of the phones.

The phones register fine, and for the most part, there is clear communication when using them, but occasionally I am getting reports that the allworx user can hear the other caller just fine, but the other caller cannot hear the allworx user.

What am I missing?

[itcoresys]

First thing, if the phones are connected through a VPN tunnel, then there are no additional holes to drill in the firewall.

The only holes in your firewall forward the server's ports for the benefit of "remote" phones out in the internet. RTP (usually udp 15000-15511) tcp 8081, udp 2088,and udp 5060. Hence the recommended usage of the Plug and Play code as an added security layer to user/password.

VPN reachable phones are "internal" not "remote".

All you need for that is to put a static route for the VPN networks in the Allworx servers with the IP address of the VPN server that goes to those remote sites.

For example:

If your Allworx server is on 10.100.0.0/24 network at 10.100.0.250, and its default gateway is the same as the VPN server, then any VPN sites need static route entries in the Allworx server pointed to the same default gateway.

Say the default gateway/VPN server is 10.100.0.1 and the remote site network of phones is 10.101.0.0/24, then the Allworx needs a static rout for 10.101.0.0/24 pointed to the same default gateway of 10.100.0.1.

After the reboot, the VPN phones will appear as internal and all your one way audio problems go away.

Warning, if possible, dont use the common private networks within your Allworx server network. If the Allworx is 192.168.0.250, 192.168.0.0/24 being an extremely common subnet, a remote phone over the internet behind a firewall with an IP of 192.168.0.x might have one way audio issues.

So, better to have your server on another random private network like 172.28.55.0/24 or 10.172.55.0/24 which is far less likely to collide with remote networks.

Finally, the point of the handset preference groups changing RTP ports and SIP UDP ports for phones is for truly remote phones behind a firewall that cannot handle multiple remote phones behind it with the same RTP and SIP UDP ports.

So if a remote (non VPN) site has 5 phones behind a firewall, then you should have 5 unique handset preferences with 5 different RTP ports and SIP ports. These same 5 could serve another completely disconnected (no VPN) remote site behind another firewall so no need for 10 handset preference groups.

Hope this helps.

[rjeffries]

I've recently gone through hell to get my remote phone working. The VPN solution was not attractive in my case, but may be fine for others.

I use Verizon DSL in California (VZ in other areas may use PPPoE, but they are DHCP here).

The configuration that finally worked (after a long series of failures) is a Westell DSL modem configured as a pure bridge,
with modem connected to a Netgear WNDR 3700v2 router.

AS we were debugging, a breakthrough of sorts came when attaching the Allworx 9112 direct to the model
would provide two way audio, everything worked. Well, that at least proved VZ was not the problem.

After that, the secret sauce is not crystal clear. A combination of three things
yielded a working remote9112 phone:

-- yet another twiddling of router settings
-- ripping out my entry at Allworx server/pbx then rebuilding (by a different expert)
-- defaulting Allworx handset and rebuilding config

Sorry the solution is unclear, but I ***finally*** have a working phone, how SWEET!

If anybody wants my router settings (only useful if you use same model) email me
rjeffries@otacomputing.com

Next step: I want to make the X-Lite SIP client work with Allworx. That would ROCK
(in addition on my 9112 handset).

[justingoldberg]

bridging probably fixed it.

You were probably double natting with the dsl mode.