[CHEINRICH]

I have followed the directions below. I have a PSTN Line running from my analog PBX into the FXO Port on the spa-3102. I still get a fast busy from the spa-921 after configuring each unit to the following specs:

Sipura back-to-back Config

This will allow the FXO connection of an SPA3000 to be accessed from another SPA unit
connected to a network somewhere. Incoming calls on the SPA3000 FXO will be
forwarded to the other Sipura unit (called the “remote SPA” in this example).
No registrations will be used, the communications will be done using IP addresses. When someone picks up the phone on the remote SPA, it will automatically connect to the FXO port on the 3102 to allow users to make normal PSTN calls.
In this example the IP Address of the SPA310 is 192.168.3.34 & the IP Address of the remote SPA is 192.168.3.35. It would also be advisable to use static addresses on both the Sipuras.

How To Setup:

In the web interface for the remote SPA in the tab for “line 1”, “EXT1” or whichever line you want to use. Alter the following settings:

–

“Make Call Without Reg” to “yes”. This allows calls to be made without being registeredto a SIP registrar.

–

“Ans Call Without Reg” to “yes”. This allows calls to be received without being
registered.

–

“Enable IP Dialing” to “yes”. This enables the SPA to dial using IP addresses rather than SIP URIs which is ideally what should be used.

–

Change the dial plan to read “(S0<:192.168.3.34>)”. This sets up what is called a “hotline”. So when the phone is picked up, it automatically dials the number (or IP address in this case) without the user doing anything.

In the web interface for the SPA3102, in the tab for “PSTN Line”. Alter the following settings:

–

“Make Call Without Reg” to “yes”.

–

“Ans Call Without Reg” to “yes”.

–

Change dialplan 2 to read “(S0<:192.168.3.35>)”. This sets up a hotline which calls the remote SPA using it's IP address. It's wise to leave use dialplan 2 (or any number after that) & not dialplan 1 since most of the default settings on this page are set to use dialplan 1.

–

“PSTN Ring Thru Line 1” if you have a phone connected to the 3102 & want it to ring when a call is received via the PSTN, leave this as “yes”. If you only want the phone on connected to the remote SPA to ring, set this to “no”.

–

“PSTN Caller Default DP “ to “2”. This should match whichever dialplan you setup two steps ago. Again, try to avoid changing dialplan 1 in this case.

–

“PSTN Answer Delay “ change this to zero. Otherwise any incoming calls will not be forwarded to the remote SPA for 16 seconds (as the default value here is 16).

In the “line 1” tab, change the following settings:

–

”make Call Without Reg” to “yes”.

–

”Ans Call Without Reg” to “yes”.
Now in the “User 1” tab alter the following setting:

–

“Cfwd All Dest” to “gw0”. This forwards any incoming calls on the VoIP line to gateway-0 which is the FXO connected to the PSTN. Now when the remote SPA calls the IP address of this 3102, it is automatically forwarded to the PSTN & the user of the remote hears the PSTN dialtone.

Is there anything wrong with this setup? Thanks for the help.

[hwittenb]

The instructions look OK.

PBXes often use different voltage levels from a PSTN line. In order for the SPA3102 to take the FXO port off hook it needs to sense that the line is not in use. It does this by measuring the voltage level on the port. The voltage level must be higher than the Line-In-Use Voltage setting. A PSTN line will have an on-hook voltage of about 48v. Off-hook the voltage drops to about 7v. The default setting for Line-In-Use is 30. You need to measure the on-hook and off-hook voltage on your PBX line attached to the 3102. You do this by reading the voltage on the SPA3102 INFO tab at the appropriate time. If your Line-in-Use setting needs to be adjusted it could be a reason for your call failure.

For the call from the SPA921 to the 3102 you are receiving the call on the 3102 Line 1 tab and forwarding the call to gw0 (VoIP-to-pstn gateway) which should give you a dial tone. Another way to do it is to receive the call on the PSTN tab. To do this you would put the pstn tab's port number in the SPA921's dial plan (S0<:192.168.3.34:5061>). It should work either way. Not mentioned in the instructions, but probably set correctly, under VoIP-to-Pstn gateway you should have VoIP-to-PSTN Gateway Enable YES, VoIP Caller Auth Method NONE, One Stage Dialing: YES, and the VoIP DP settings set to number 1 which is (xx.).

For the call from the PBX to the 3102 to the SPA921 this goes from the pstn-to-VoIP gateway and the instructions say to setup the hotline dial plan to 192.168.3.34 (goes to the default 5060 port number unless another port number is specified) and setup this dial plan as the PSTN Caller Default DP. Not mentioned but probably set correctly under Pstn-to-VoIP Gateway you should have PSTN-to-VoIP Gateway Enable YES, PSTN Caller Auth Method NONE, Off Hook While Calling VoIP NO.

The codex setting should say the preferred codec is G711u or use preferred codec only NO.

The hot-line dial plans you are using assume that you have blank (i.e. empty) userid fields on each device tab.

There is also an alternative way to configure the SPA921 and the SPA3102 to dial the digits directly on the SPA921 and pass them to the SPA3102 to dial on the pstn-to-VoIP gateway. This is an alternative to getting the dial tone from the SPA3102.

[humba3]

If you compare this with my SPA9k SPA3k interop guide, you'll see quite a bit of difference in the DP.. e.g. S0 should go at the end of the DP, and your substitution sequence should contain userid, and port number. The port number is quite important to make sure the call goes to the proper line.

So e.g. if you configured the PSTN Line sip userid to 123, the DP on the phone should be

<:123@192.168.3.34:5061>

Likewise, when you send calls to the phone.. it has to be <:userid-spa9xx@192.168.3.35:5060> (you might be able to leave out the port here because you're using the default port).
Naturally, line 1 also needs to be enabled, but registration needs to be turned off.

Bottom line, check my interop guide, the IP dialing scenario can be applied 1:1 to a phone <-> gateway scenario.

Also.. it's quite important that you use the userid.. without it, it's like you just put an address on a package you're shipping out. If you add the port, that's like adding the floor, but often, you need the name for the package to arrive at the proper destination. The calls you send to your spa3k will, if at all, go to the VoIP1 line and that's definitely not where you want your outgoing calls to end up.