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Cisco/Linksys (Sipura) VoIP Support Forum Technical support, how-to guides, troubleshooting, and general assistance for Cisco and Linksys VoIP hardware, including the highly recommended SPA3102 Voice Gateway and the SPA525G IP telephone.



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  #1 (permalink)  
Old February 27th, 2009, 03:21 PM
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Default SPA3102 Features and Call Flow

I just bought a SPA3102 and after some reading found out that it is a lot more powerful that I had initially thought.

I have also realized that alot of people are not aware of what it can and cannot do, and that a lot of the information is spread out over thousands of posts on many different forums.

I would like to clarify some of my own questions and hopefully edit the first (this) post to summarize my findings.



What I understand so far.



The SPA3102 has two parts to it:
1. PSTN-SIP gateway (call this the Gateway) -- PSTN Tab
2. ATA -- Line1 Tab

- The Gateway (PSTN tab) has SIP on one side and FXO (landline) on the other.
- The ATA (Line1 tab) has SIP on one side and FXS (telephone) on the other.

FXS = telephone jack
FXO = line jack

- Both the Gateway and the ATA can have individually registered SIP accounts, thereby allowing each to answer an incoming SIP call.

Question: Can they both the ATA and Gateway answer incoming SIP calls at the same time?



Incoming SIP calls
- Gateway SIP --> FXO
- ATA SIP --> FXS or FXO

Question: Can incoming Gateway SIP be forwarded out over ATA SIP?
Question: Can incoming ATA SIP be forwarded out over Gateway SIP?

Incoming FXO calls
- FXO --> Gateway SIP or FXS
- FXS --> FXO or ATA SIP

Question: Can incoming FXO be forwarded out over ATA SIP?
Question: Can a call be made from the FXS be forwarded out over the Gateway SIP?


I will move on to dial plans and gateways next.

Last edited by zalnn; February 27th, 2009 at 04:00 PM.
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Old February 27th, 2009, 05:31 PM
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Default Re: SPA3102 Features and Call Flow

Quote:
Originally Posted by zalnn View Post
Question: Can they both the ATA and Gateway answer incoming SIP calls at the same time?
Yes


Quote:
Incoming SIP calls
- Gateway SIP --> FXO
- ATA SIP --> FXS or FXO

Question: Can incoming Gateway SIP be forwarded out over ATA SIP?
Question: Can incoming ATA SIP be forwarded out over Gateway SIP?
No to both. You can not make a sip to sip call.

Quote:
Incoming FXO calls
- FXO --> Gateway SIP or FXS
- FXS --> FXO or ATA SIP

Question: Can incoming FXO be forwarded out over ATA SIP?
Question: Can a call be made from the FXS be forwarded out over the Gateway SIP?
An incoming call from the FXO port can not be forwarded out over Line 1 Sip but with an appropriate dial plan it can use credentials setup under one of the four gateway accounts on the Line 1 tab. If you also setup the Line 1 configuration under one of the gateways the call will go out using the basic parameters setup on the PSTN Line tab.

A call from the phone attached to the FXS port can be directed with a dial plan to go out over the FXO port.

You may wish to review the ATA Administration Guide for technical details about the SPA3102.

Last edited by hwittenb; February 27th, 2009 at 05:33 PM.
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Old March 2nd, 2009, 07:31 PM
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Default Re: SPA3102 Features and Call Flow

thanks... that cleared up a lot for me!

IF you want to create a SIP to SIP call, there is a hack.
All you have to do is connect the phone jack to the line jack using a regular phone cable (RJ-11 ??)
Then configure the PSTN to answer and forward.

Last edited by zalnn; March 2nd, 2009 at 07:35 PM.
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Old April 15th, 2009, 05:34 PM
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Default Re: SPA3102 Features and Call Flow

can an incoming call on the SIP account registered on the PSTN page/tab ring the phone (FXS)?
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Old April 15th, 2009, 05:43 PM
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Default Re: SPA3102 Features and Call Flow

Quote:
Originally Posted by zalnn View Post
can an incoming call on the SIP account registered on the PSTN page/tab ring the phone (FXS)?
No it cannot ring the phone attached to the "Phone" (FXS) port. An incoming call on the SIP account can be bridged to an outgoing call on the FXO port.
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Old April 23rd, 2009, 04:09 AM
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Default Re: SPA3102 Features and Call Flow

I have a Milkfish SIP (DD-WRT) server in one building connected to a Cablemodem, that is equipped with a working FXS telephone port. This router is connected to a second building.

If I purchase a (remote) SIP phone for building 2, can a SPA3102 bridge the SIP phone (of building to) to the FXS RJ11 telephone port of the cable modem?

I would like to use the remote SIP phone to acquire control over the FXO port, so that it can make outgoing PSTN calls through the cable modem's FXS telephone. Is this the right device for the job? Any success stories are appreciated. Thank you.
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Old April 23rd, 2009, 03:59 PM
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Default Re: SPA3102 Features and Call Flow

Quote:
Originally Posted by gatorback View Post
I have a Milkfish SIP (DD-WRT)
If I purchase a (remote) SIP phone for building 2, can a SPA3102 bridge the SIP phone (of building to) to the FXS RJ11 telephone port of the cable modem?

I would like to use the remote SIP phone to acquire control over the FXO port, so that it can make outgoing PSTN calls through the cable modem's FXS telephone.
Yes, the SPA3102 can bridge a call from the remote sip phone to the telephone line (cable phone) attached to the FXO port of the SPA3102.

I am not sure about the impact of the Milkfish server, as I don't have any experience with that, but it potentially could make things easier. You can brigde the calls without the the Milkfish server.
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