Setting up a VoIP extension on a local network.

Question

With the help of the experts on this board I have successfully set up a VoIP phone extension on our private network. The questions & answers can be viewed at. http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&thread.id=3197 . For the benefit of anyone attempting a similar project, here is the completed setup.
This installation is in a small motel in Te Anau, on New Zealand’s South island. The manager lives off site, and needs to be able to receive calls at night, and also transfer incoming calls to guest’s extensions through the hotels PBX. This necessitates a direct link to the PBX, rather than simply diverting the phone. One solution would have been to lease a circuit from the local Telco, but in NZ, this is very expensive, so another solution was sought. Fortunately there was an established wireless data link between the hotel and the managers residence, so VoIP seemed the obvious choice.
The equipment used is a Linksys SPA3102 connected to an extension on the PBX, and a Linksys PAP2 at the remote end. The setup would work equally well if connected to a phone line, rather than the PBX.
I’ll start the setup with the SPA3102.
Connect the POTS line to the LINE port, and your switch/router to the INTERNET port. In my setup the Ethernet port is not used. Plug a standard phone into the Phone port. This is useful for testing and setting up. It’s not needed afterwards, unless you want a local phone.
Open your web browser, and type the adaptor IP into the address bar. Go to Admin, and Advanced Settings.
ROUTER SETUP
WAN Setup Tab:
Connection type: Static IP.
Static IP Settings: The Network address on your local network (192.168.x.x)
Subnet mask 255.255.255.0
LAN Setup Tab:
LAN IP address: This is automatically selected to be on a different sub net from the WAN. Unless it conflicts with another address on your system you shouldn’t change it.
Enable DHCP: No
(Save these settings.)
VOICE SETUP
System Tab: No Changes
SIP Tab: No Changes
Provisioning Tab: No Changes
Regional Tab: Mostly this sets the dial tones etc to match your local service. Unless you need them to be the same this shouldn’t need any changes
The Hook Flash Timer Min & Max: should be set to the local values. The Defaults (.1 and .9) are OK for North America. Australia and New Zealand use .07 & .13. If you have trouble sending a hook flash, check these values against the local settings.
DTMF playback level should be greater than zero. (I used 3)
(Save these settings)
Line 1 Tab:
I don’t use Line 1 except for testing. During setup the line should be enabled. After the system is running OK, it can be disabled
Line enabled yes
SIP port 5060
Proxies are not used in this setup.
Register: No
Make call without reg: yes
Answer call without reg: yes
User ID: 10? (you can use any number)
Line 1 Tabupplementary services.
Change Call waiting, 3 way Conf, and 3 way call, to no. (These interfere with sending a hook flash)
Hook Flash Tx method: AVT
(save these settings)
PSTN Line Tab
Line enable: yes
SIP Port 5061 (default)
Proxy: proxies are not used.
Register: no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like (VoIP gateway?)
User ID: leave blank
User password: leave blank
Use auth ID: no
Dial Plan 1: (<:*>S0). Switches to the outside line when * received.
Dial Plan 2: (<:[email protected]:5060>S0). 11 is the user ID on the PAP2
VoIP to PSTN enable: yes
VoIP caller default DP: 1
One stage Dialing: no
VoIP users & Passwords.
User 1 ID: 11. User1 DP: 1
User 2 ID: 21 User 2 DP: 1
User 3 ID 22 User 3 DP: 1
(These are the line numbers of additional PAP2’s on our system)
PSTN to VOIP Gateway enable: yes
PSTN Caller ID none
PSTN Caller Default DP: 2
Detect PSTN long silence yes
Detect VoIP long silence yes
Detect Disconnect tone yes
VoIP answer delay 0
PSTN Answer delay 0
PSTN to VoIP gain (Set these to adjust
VoIP to PSTN gain the speech volume)
Line in Use voltage: This should be set midway between the On Hook and Off Hook voltages, which you get from the Info screen. Most public phones are 47v on hook, and 7v off hook, so the setting should be 27v. My PBX is 27v on hook, and 7v off hook, so my setting is 17v. To read this, go to the Info screen and check the Line Voltage, then go Off hook (make a call), click the reload button on your browser, and check the line voltage again.
(save these settings)
This completes the setting up of the SPA3102.
Now for the setup of the PAP2.
Open your web browser, and type the PAP2 IP into the address bar. Go to Admin, and Advanced Settings.
System tab:
DHCP no
Static IP 192.168.x.x (same sub-net as your network. Different adaptor number)
Net Mask 255.255.255.0
(save these settings)
SIP Tab: no changes.
Provisioning Tab: no Changes
Regional Tab.
Hook Flash Min & Max: change to your local settings if required.
(save these settings)
Line 1 & Line 2 Tabs.
Whether you use Line 2 depends on whether you want to have 2 phones on the PAP2. All calls from the PSTN line of the SPA3102 will go to Line 1 of the PAP2 as per Dial Plan 2 on the SPA
Line enable yes
SIP port 5060 (line 1) & 5061 (line 2)
Proxy Proxies are not used.
Register no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like
User ID 11 (line 1) & 12 (line 2)
(These are used to identify each line on the system)
Call waiting: no
3 way conf: no
3 way call: no
DTMF Tx method: AVT
Dial Plan: This is the dial plan I use on line 1.
(<:192.168.4.10:5061>S3|21S0<:@192.168.4.9:5060>|22S0<:@192.168.4.9:5061>)
You will have to modify it for use on other lines, or other adaptors, and the IP addresses must match your system IP addresses. Here is an explanation.
192.168.4.10:50613 All my adaptors are on subnet 4. 10 is the number of the SPA3102, and 5061 is the SIP port mapped to the PSTN line. If the handset is lifted, and no numbers are dialed the call will be transferred to the PSTN line after 3 seconds, and you will hear the outside dial tone. If within 3 seconds you dial either 21, or 22, the phone on either line 1, SIP port 5060, or Line 2, SIP port 5061, on adaptor 9 will ring. (If you only have one PAP2 then you will only need the first section of this dial plan.)
Enable IP Dialing: yes
(save these settings).
User 1 and User 2 tabs: no changes
That just about does it. All incoming calls from outside are received by the PBX, and after hours are sent to the extension connected to the SPA3102, which rings the phone on the remote PAP2 in the manager’s house. If the call is for a guest we can press the recall button (hook flash), dial the guest’s extension number, and transfer the call when they answer. As an added bonus we have a second PAP2 elsewhere on the network, and we can call between the 3 adaptors. All 3 adaptors have access to an outside line, though the PBX. I’m fairly sure it would also work through a VPN, which would mean we could take a VoIP phone anywhere in the world, and still be virtually ‘On site". I don’t know if that is a good thing or not.

Answer 1

Hi HW,
The PBX is a Panasonic TA308. There is no special interface to the PBX,  the  line port on the SPA3102 is simply plugged into an extension, like another phone. Anyone calling that extension will have the call routed through the SPA & PAP2 to the remote phone.
The whole setup is totally seamless, & transparent to the user. As we are on a local network there is virtually no latency. There is a slight tendancy to echo,  but the echo suppression mostly takes care of that.
THis has been a good exercise, and once I got my head around what I was trying to do, with your help,  it was pretty easy.  I think the hook flash timing would be the thing which gives most users a problem, as it seems to vary widely around the world. I was surprised at the difference between the US and NZ (.1 & .9 to .07 & .13).  There didn't seem to be any other critical differences.
Now I am the local expert on VoIP   "In the Kingdom of the blind, the one-eyed man is King."