[PhoneBoy]

I've actually had a unit for a few weeks, so I've gotten to use it and can give you an idea of how you can use it.

There are two lines on the SPA3000: "Line 1" which is a standard SIP line, and "PSTN Line" which is kind of both SIP and PSTN. You can plug a real PSTN line into the PSTN Line, or you can plug in a locked device from a provider such as VoicePulse, Broadvox, Broadvoice, etc. Line 1 functions more or less like a SPA-2000 line except it has some additional call routing options, which I will get to in a moment.

If you look in the configuration for the PSTN Line, you'll notice it contains SIP information as well. This is used for two purposes:

1. PSTN to VoIP bridging: You can call in from a PSTN line and, depending on how you set up the authentication, you'll be able to make a SIP call. You can restrict what the SIP call will be by dialplan. A number of different PINs can be defined, each with a different dial plan.

2. VoIP to PSTN bridging: You call in remotely via SIP, authenticate (PIN and HTTP Digest are supported), and then you can make a call on the PSTN line. Again, each PIN can be restricted to a specific dial plan.

3. PSTN Fallback: If the SPA3000 loses power, Line 1 will essentially "pass thru" straight to the PSTN line, thus allowing you to make calls. If power is restored, any in-progress call will still be maintained. This is configurable (I believe it is off by default)

4. PSTN Calls can ring on "Line 1" automatically (configurable). This essentially allows you to route incoming calls via SIP and PSTN to the handset hooked up to line 1. With the dial plan, you can also make calls on SIP and PSTN from the same handset (see below).

Line 1 is basically like a line on a SPA-1000/2000 except that there are additional call routing options. In addition to the primary line settings, up to four different "gateways" can be defined to make outgoing calls. You essentially choose which gateway to use by dialing plan. This allows you to make outgoing calls on a number of different providers.

To show you how this works, I will share my configuration with you.

In my case, Line 1 is configured for one of my Free World Dialup (FWD) numbers as primary, the following as gateways:

gw1: VoicePulse Connect line
gw2: iConnectHere
gw3: SIPphone

In the dialplan, there is also a gw0, which refers to the PSTN Line.

I've used a BroadVoice and a Broadvox Direct SPA-2000 as a PSTN Line. I've also done brief tests with a real PSTN Line from Qwest.

Dial plan is basically:

Regular dialing for PSTN
Dial #3 + number for FWD
Dial #8 + number for VoicePulse
Dial #4 + number for iConnectHere
Dial #7 + number for SIPphone

The dial plan string is:

(<#3,:>xx.|<#3,:>*xx.|<#3,:>**xx.|<#8,:>xx.<:@gw1> |<#4,:>xx.<:@gw2>|<#7,:>xx.<:@gw3>|xx.<:@gw0>)

Actually, the default dialing for PSTN only works if I append a "pound" after dialing the number, otherwise it will use FWD to dial the number. That's a bug I've already reported to Sipura.

An alternate dialplan, which is basically the same as above except #9 will be used to make a PSTN call and FWD will be used otherwise, is:

(<#9,:>xx.<:@gw0>|<#8,:>xx.<:@gw1>|<#4,:>xx.<:@gw2 >|<#7,:>xx.<:@gw3>|xx.|*xx.|**xx.)

One new feature is Call Forward Busy and Call Forward No Answer (defined on the User 1 tab). You can also forward calls to the PSTN for all/busy/no answer calls using the syntax: number@gw0. Haven't tested this yet.

Here are some issues I've discovered (all of these reported to Sipura already):

1. Gateway passwords (e.g. Gateway 1, Gateway 2, etc) should be obscured.

2. Should be able to configure Caller ID information for each Gateway.
Currently, outgoing Caller ID via VoicePulse shows my FWD number or
"unavailable" depending on who I call. Outbound Caller ID on
iConnectHere shows a random number, which is normal for iConnectHere.

3. Inbound calls from the PSTN line ring through via Line 1 with a 5 second delay. This is actually expected behavior as the Caller ID information isn't passed on a US PSTN line until between the first and second ring. SIP INVITEs need to pass Caller ID before the phone rings. The five second delay is needed to capture the Caller ID information. Otherwise, the SIP information from the "PSTN Line" configuration may be substituted.

4. Inbound Calling to the PSTN line may not always pass Caller ID
information consistently. It will either pass nothing or it will pass the FWD
configuration from my PSTN Line configuration, which is of course not
what I expect. Sipura is working on it.

[sk3-403]

Originally Posted by PhoneBoy

3. Inbound calls from the PSTN line ring through via Line 1 with a 5 second delay. This is actually expected behavior as the Caller ID information isn't passed on a US PSTN line until between the first and second ring. SIP INVITEs need to pass Caller ID before the phone rings. The five second delay is needed to capture the Caller ID information. Otherwise, the SIP information from the "PSTN Line" configuration may be substituted.

This is a really nasty, nasty thing.

I've also noticed that as well as taking 5 seconds before it starts ringing, it takes the same amount of time after the call is terminated. So the Phone attached to your Sipura will still be ringing, even though the PSTN call stopped a while ago.

This could be related to not correctly detecting disconnection though, as mentioned elsewhere.

[PhoneBoy]

I believe it takes the Sipura up to 8s to detect no ringtone. Sipura said this could definately be optimized.

[sk3-204]

Have you managed to get PSTN to VoIP to work?

I have a SIP endpoint at 5917 off the proxy to which the 3000 is registered.

I have PSTN-to-VoIP PSTN Caller default DP = 1
Dial plan 1 reads: (<:5917>) [I am told that (<x.:5917>) currently has a problem
PSTN access list = *
PSTN Caller ID pattern = *
PSTN line shows it is registered with the proxy ok.

Using Ethereal I see no SIP activity;
Syslog (debug option 1-line) indicates PSTN call ringing and later ending, but no SIP or other SYSlog messages occur in between.

Hope someone has found a solution

[sk3-403]

I've had PSTN to SIP working by means of forwarding all calls to an alternate address (I did this to redirect calls coming in on my PSTN line to my Cisco 7960)

[PhoneBoy]

Here's another sample configuration to get you thinking.

Line 1 is connected to an Asterisk box at Voxilla. Gateways are set up to VoicePulse (gw1), iConnectHere (gw2), SIPphone (gw3), and FWD (gw4).

PSTN Line is connected to a real Qwest PSTN line. SIP Configuration on the PSTN line is configured for FWD.

An inbound call to my Voxilla extension, FWD number, or my PSTN line will generate a ring on one line of my phone. The PSTN line will ring through with a five second delay of course. The trick to accomplishing this is to either set the VoIP-To-PSTN Gateway Enable to No or to set the VoIP Caller Auth Method to HTTP Digest, depending on your needs.

I can, of course, make outbound calling on any of the gateways with the appropriate dialplan. #3 for an FWD call, #8 for a VoicePulse call, #4 for an iConnectHere call, #7 for a SIPphone call, #* for a call through the Voxilla Asterisk box, all else goes out the PSTN line.

(<#3,:>xx.<:@gw4>|<#3,:>*xx.<:@gw4>|<#3,:>**xx.<:@ gw4>|<#8,:>xx.<:@gw1>|<#4,:>xx.<:@gw2>| <#7,:>xx.<:@gw3>|<#*,:>xx.|xx.<:@gw0>|*xx.<:@gw0 >)

[scanchain]

Hi PhoneBoy,

I tried to follow your example to get my phone connected to Line1 to ring.

My setup is as follows: (a) FWD on Line1, (b) Stanaphone on SIP configuration under PSTN and (c) landline is connected to the PSTN port.

When I dial into my Stanaphone number, I could never get the phone to ring. (I have verified that the "Registration Status" shows "Registered" under the "PSTN Line Status")

I tried

1) Setting "VoIP-To-PSTN Gateway" to Enable, in which case, I got a series of beeps when I dial into the Stanaphone number. I interpret these as the prompt to enter pin for the "VoIP-To-PSTN Gateway" feature.

2) Setting "VoIP-To-PSTN Gateway" to Off, in which case I received a "Number is not available" message when I dial into the Stanaphone number.

I suspect a "Ring Thru Line 1" check-box is missing in the "VoIP-To-PSTN Gateway Setup" section. I only have the "PSTN Ring Thru Line 1" check-box under "PSTN-To-VoIP Gateway Setup" section.

Do you have any idea why the phone does not ring when I dial into the Stanaphone number?

Thanks!

[PhoneBoy]

Well I did mean "PSTN Ring Thru Line 1" (I just didn't have the option in front of me). Try leaving VoIP to PSTN Gateway enabled and either:

1) Set the VoIP Answer Delay really high (like 30 seconds)
2) Enable HTTP Digest authentication (which I use to connect my PSTN line to Asterisk)

[scanchain]

Hi PhoneBoy, I have another question regarding the gateway configuration.

Currently, I have Line1 configured to FWD. In addition to that, I want to be able to dial out using SIPphone. My dial plan is

(<:*1408>[2-9]xxxxxx|<:*1>[2-9]xx[2-9]xxxxxx|<1:*1>[2-9]xx[2-9]xxxxxx|<#0,:>xx.<:@gw0>| <#1,:>xx.|<#1,:>*xx.|<#1,:>**xx.|<#2,:>xx.<:@gw2>| <#3,:>xx.<:@gw3>|011xxxxxxx.<:@gw0>)

where <#2,:>xx.<:@gw2> is for dialing out using SIPphone. I have gateway 2 setup as follows:

Gateway 2: proxy01.sipphone.com
GW2 NAT Mapping Enable: no
GW2 Auth ID: 1747xxxxxxx (where actual digits were replaced by 'x')
GW2 Password: <my password>

However, I have not been able to dial out. Trying the SIPphone welcome recording number (by dialing #217474745000) results in an invalid number tone.

I hope you can advise me if I missed some settings. Do I have to enable NAT or add STUN settings (these are at factory defaults) or open up ports on my router?

Thanks a lot!

[PhoneBoy]

I think you need to enable NAT Mapping to dial out via SIPphone and/or configure STUN in the SIP tab.