PSTN - SPA3012 - LAN - SPA3012 - Analogue Phone

[TricksDrummer]

Hi there,
Well after checking out a couple of forums and not having success with setting up the SPA3012's i thought i'd ask nicely if someone would clarify the settings required to achieve the following, i have also reset both units to factory default and and then changed the LAN ip's to match my LAN, also disabled DHCP.
I have the following setup:
Unit1:
IP: 192.168.1.21
Phone line plugged into "Line" port
Analogue phone plugged into "Phone" port(phone1)
Network cable plugged into "Internet" port

Unit2:
IP: 192.168.1.23
Analogue phone plugged into "Phone" port(phone2)
Network cable plugged into "Internet" port

How i would like it configured is:
If phone1 is picked up they can dial out normally through the phone line plugged into Unit1.
If phone2 is picked up they can dial out normally through the phone line plugged into Unit1.
If an outside call comes in on the phone line plugged into Unit1, only phone1 will ring.
=Optional(But preferred)========================
If "#1" is pressed on phone1, it will call phone2
If "#1" is pressed on phone2, it will call phone1
Block caller ID on calls made from phone2
==========================================

Im in New Zealand if that helps...(less of the sheep shagger jokes )

[hwittenb]

The assumption is both units are on the same local lan.

There are a couple of ways to configure. You either configure to get a dial tone from the distant ata for the outgoing call and then key to send over the lan the dialing digits to the distant ata or you configure to send the sip invite to the distant ata with the number and distant ata dials the number. This configuration is the latter way. The former way is more common.

Unit 1
Unit 1 Line 1 Tab
Sip Port: 5060
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
UserID: Unit1Line
Dial Plan (<#1:>S0<:Unit2Line@192.168.1.23:5090>|xx.<:@gw0> )
Enable IP Dialing: Yes

Unit 1 PSTN Line Tab:
Sip Port: 5061
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
Register: No
UserID: Unit1pstn
VoIP-to-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: none
One Stage Dialing: Yes
RingThru Line 1: Yes
PSTN Answer Delay: 45 (You don't want an incoming pstn line call answered by the Unit 1 pstn-to-voip gateway, you just want it to ring-thru to the phone attached to Line 1)

Unit 2:
Unit 2 Line 1 Tab:
Sip Port: 5090
Make Call Without Reg: Yes
Ans Call Without Reg: Yes
Register No
Proxy: 192.168.1.21:5061
UserID: Unit2Line
Dial Plan (<#1:>S0<:Unit1Line@192.168.1.21:5060>|<:0197>xx. )
Enable IP Dialing: Yes


Phone 2 dials out thru the phone line plugged into Unit 1 FXO port. To block caller id you need to send something to your telephone company. You would append that code to the calls going out. A Google internet search indicates that you do this in New Zealand by appending 0197 to the call. If this is not the way you do it adjust the dial plan. In the above all calls from Unit 2 Line 1 have 0197 appended to the call (except the call that goes to the opposite unit's phone set).

The Unit 2 PSTN Line tab sets up the proxy as the pstn line tab on unit 1. This causes outgoing voip calls to be sent to that address. The phone number sent in the voip call does not match the user id on the pstn line tab so Unit 1 dials the call on the FXO port.

To change the configuration to configure where you get the dial tone from the distant unit and then key the digits to be dialed, you would make the following changes to Unit 2:

Line 1 Dial Plan: Dial Plan (<#1:>S0<:Unit1Line@192.168.1.21:5060>| <#2:>S0<Unit1pstn@192.168.1.21:5061>)
And then on Unit 1 you would need to adjust the VoIP Caller Default DP using one of the other numbers to append the caller id blocking to the number dialed. In this case you need to also dial something to differentiate the call from the call where you are dialing the other ata's handset. I setup #2 for that.

[TricksDrummer]

You Rock!
Worked first time like a charm
that option "register:no" was new to me.

1 last request,
If a call comes in to Unit1 from the phone line plugged into the "Line" port,
And Phone1 Answers it (as it should and only can),
Can i set an option to be able to transfer it from Phone1 to Phone2?
And/or create a conference call with Phone2, Phone1 and phone line?

Thanks so much

[hwittenb]

Originally Posted by TricksDrummer

If a call comes in to Unit1 from the phone line plugged into the "Line" port,
And Phone1 Answers it (as it should and only can),
Can i set an option to be able to transfer it from Phone1 to Phone2?
And/or create a conference call with Phone2, Phone1 and phone line?

Thanks so much

I'm glad to know that it worked! The "register: no" setting was really only required on Unit2 where you had an entry under Proxy. I suggested it on the others just for good measure.

As for the transfer/conference calling question, you might be able to. I don't have a good way to test it. You would use the SPA's internal supplementary service feature signalled by a flash. For dialing I would try someting in the dial plan, an additional element like |<#2:Unit2Line@192.168.1.123:5090>S0| or maybe put a single digit element like |x| and put the sip url in a speed dial element (on the User1 Tab). If you put it in speed dial 3 you would dial 3#.

[TricksDrummer]

Originally Posted by hwittenb

You would use the SPA's internal supplementary service feature signalled by a flash. For dialing I would try someting in the dial plan, an additional element like |<#2:Unit2Line@192.168.1.123:5090>S0|

I put the into the dial plan, but im not sure what you mean about internal supplementary service feature signaled by a flash, is there a setting somewhere i need to enable/set?

[hwittenb]

Originally Posted by TricksDrummer

I put the into the dial plan, but im not sure what you mean about internal supplementary service feature signaled by a flash, is there a setting somewhere i need to enable/set?

It is set by default, one of the "supplementary service" features listed on the Line 1 tab. It is possible to disable the feature and for some applications you need to disable it.

Here is what one of the older ATA Administration Guides says about Call Transfer and Three Way Calling. The description isn't in the latest Guide. Why they took it out on the rewrite, I don't know. Maybe they didn't want to answer questions about it.

Attendant Call Transfer
Service description
Attendant Call Transfer lets a customer use their touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones.
User action required to activate or use
While in a call with the party to be transferred:
1.Press the switch hook or flash button on the phone to place the party on hold
2.Listen for three short tones followed by dial tone
3.Dial the number to which you will transfer the caller
4.Stay on the line until the called number answers
5.Announce the call
6.Press the switch hook or flash button adding the held party to the call
7.Hang up to connect the two parties and transfer the call
NoteYou can hook flash while the third party is ringing to start an early conference, and then hang up to complete the transfer without waiting for the third party to answer first.

Expected call and network behavior
When the user presses the switch hook or flash button, the transferee is placed on hold. When the user successfully dials the transfer number and the party answers, the transferee can be added to the call by pressing the switch hook or flash button, creating a three-way conference. When the user hangs up the phone, the transferee and the called party remain in a call.

User action required to deactivate or end
Not applicable.
Service description

Unattended or “Blind” Call Transfer lets a customer use their touchtone phone to send a call to any other phone, inside or outside their business, including a wireless phones.
User action required to activate or use
While in a call with the party to be transferred:
1.Press the switch hook or flash button on the phone to place the party on hold
2.Enter *98
3.Dial the number to which you will transfer the caller
The call is transferred when a complete number is entered. You will hear a short confirmation tone, followed by regular dial tone

Call Hold
Expected call and network behavior
When the user presses the switch hook or flash button, the transferee is placed on hold. When the user successfully dials the transfer number, the transferee automatically calls the dialed number.
User action required to deactivate or end
Not applicable.

Service description
Call Hold lets you put a caller on hold for an unlimited period of time. It is especially useful on phones without the hold button. Unlike a hold button, this feature provides access to a dial tone while the call is being held.
User action required to activate or use
Press the switch hook or flash button on the phone to place the first party on hold. You will hear a dial tone.
To make another call, enter the new number.
To return to call on hold, hang up, and the phone set rings with the first call on the line (or hook flash again)
Expected call and network behavior
User action required to deactivate or end
Hang up the telephone.

Three Way Calling
Service description
The user can originate a call to a third party while engaging in an active call.
User action required to activate or use
1.Press the switch hook or flash button on the phone to place the first party on hold
2.Listen for three short tones followed by dial tone
3.Dial the number of the third party.
4.When the third party answers you may have a conversation with them while the other party is on hold.
To hold a conference with the party on hold and the third party, simply press the switch hook or flash button.
Expected call and network behavior
The SPA supports up to two calls per line. The SPA can conference two calls by bridging the second and third parties.
User action required to deactivate or end
Hang up the telephone.