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remote phone provisioning

Posted by MikeWitlin on Tue, 10/19/2010

I have a ThirdLane (single tenant) server running with dual NICs -

The 1st Nic is bound to internal subnet behind firewall and all internal phones (Polycom IP550) provision perfectly and work flawlessly.

The 2nd NIC is bound to a public IP (ISP is next hop). I am trying to use IP550 from remote locations (users homes). I can get the phones to provision just fine but they are unable to make any calls (URL calls disabled).

I checkd the Linux firewall settings and the SIP ports (5060; 10000-20000) are all set to allow. I have another IP550 working with a different hosted provider at the remote location and it works great.

I'm not sure what I'm missing here but I sure could use any suggestions.

Thanks


Submitted by mattdarnell on Tue, 10/19/2010 Permalink

I would eliminate the firewall as the issue.
For a short time turn it off, see if the problem goes away, turn it back on.
Then go from there.

Other possible issues:
is asterisk bound to both NICs?
what is the IP in the config file, inside or outside?

-Matt

Submitted by MikeWitlin on Fri, 10/22/2010 Permalink

Thanks - it was the variable in provisioning templates ($SERVER).

I changed it under System Settigns -> Provisioning Settings from inside IP to outside IP - reboot phone an all is good.

Submitted by eeman on Sat, 10/23/2010 Permalink

in situations like this you might want to use a little DNS magic... pick a FQDN, which on the internet resolves to your public IP. then internally make sure the responses for that same FQDN resolve for the internal IP.

Submitted by virtuallyanywhere on Wed, 02/23/2011 Permalink

I have installed a thirdlane multi-tenant system. the PRI inconfigured and phones inside the local network can call between each other and recive and make calls via the PRI. I have set up some remote phones and although the remote phones can receive and make calls vai the PRI, there is no audio.

I have configured sip_nat.conf to show the external and internal networks
rtp.conf has the port range from 10000 to 20000

the netgear router appears to be configured to forward all the appropriate ports to the ip pbx but no audio on remote phones makes me believe the issue is in the router.

the pbx is configured with a single NIC connected to one of the LAN ports of the router. the customer's internal network is connected to one of the other LAN ports

anyone who could help would be appreciated.

Submitted by eeman on Wed, 02/23/2011 Permalink

ok first of all there is no sip_nat.conf for thirdlane.. so is this really thirdlane? Trying to make a sip_nat.conf for thirdlane is the equivalent of masturbating with a cheese grater. It serves no purpose and you are not going to get the results you were expecting.

secondly is this thirdlane box behind NAT? My first requirement is to get rid of NAT entirely and put another interface on your PBX with a public IP. If you do not have enough NICs to do this then you should use an 802.1Q capable switch and use VLAN tagging on your PBX to create the extra interfaces. You will be surprised exactly how much problems NAT can create for you.

Submitted by virtuallyanywhere on Wed, 02/23/2011 Permalink

ok fine. so I can either configure the NIC as a public interface but then I need to configure iptables to protect the server. i can remove the settings for NAT. Can I set up the router to forward all relevant traffic to the ip pbx and still use it as a firewall?

Submitted by eeman on Wed, 02/23/2011 Permalink

thirdlane already had iptables running if you installed from the iso. Your router is going to do lots of nasty things including tearing down connections in its state table it determined 'expired'. You'll save yourself a lot of one-way audio nightmares.