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Issue connecting SPA3102 FXO port to paging interface

kyle
Level 1
Level 1

Hello,

I was referred to this community after a call to small business support, we were unable to come to a solution.

SPA3102 v5.2.13 FW

Viking Electronics PI-1 Paging Interface

We were told from Viking that the interface only supports an FXO port, all the connections were verified with their support and we're able to page by connecting an analog phone straight to the interface. Plugging into the FXS port on the SPA3102 also results in a dial tone playing over the paging system.

All settings on the SPA3102 are default minus SIP registration information, we see the unit as registered within the web interface as well as within our Asterisk CLI.

Trying to ring the extension results in constant ringing if voicemail is disabled, and straight to voicemail if enabled. Within the Asterisk CLI we see a SIP response 503 "Service Unavailable."

I can get the SPA3102 to respond by lowering the "Line In Use Voltage" down from its default value of 30. However all I hear is a hum and I'm not able to speak over the paging system.

Any insight into this issue would be greatly appreciated, we believe it's just settings on the SPA3102 unit itself that need to be changed.

Thanks!

1 Accepted Solution

Accepted Solutions

Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE.  The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.

I would try the following:

Viking Pl-1

Talk Battery Switch (PT): On

Audio In: Cabled to FXO port of SPA3102

SPA3102:

Interface Analog Phone attached to SPA3102:

Line 1 Tab

Line Enable: YES

Dial Plan: (S0<:@gw0>)

PSTN Line Tab

Line Enable: Yes

VoIP-to-PSTN Gateway Enable: Yes

One Stage Dialing: Yes

Line 1 VoIP Caller DP: NONE

VoIP Answer Delay: 0

Line In Use Voltage: xx

The Line In Use Voltage needs to be lower than the Talk Battery voltage supplied by the Viking Pl-1.  Usually it is set about half way between the on-hook and off-hook voltage level.  Measure the FXO (PSTN Line) on-hook voltage by reading it on the SPA3102 INFO Tab.  Set the Line In Use Voltage substantially below the on-hook talk voltage.

If you lift the phone with the above settings you should be connected to the paging system.

Interface to an asterisk pbx system:

PSTN Line Tab

Setup Registration on PSTN Line Tab to Asterisk system. 

Register: Yes

Proxy: xxx

UserID: xxx

Password: xxx

VoIP-to-PSTN Gateway Enable: Yes

VoIP Caller Auth Method: None

One Stage Dialing: Yes

VoIP Caller Default DP: NONE

VoIP Answer Delay: 0

Line In Use Voltage: same comments as above

If you call the extension on the asterisk PBX you should be attached to the Viking unit.

View solution in original post

2 Replies 2

Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE.  The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.

I would try the following:

Viking Pl-1

Talk Battery Switch (PT): On

Audio In: Cabled to FXO port of SPA3102

SPA3102:

Interface Analog Phone attached to SPA3102:

Line 1 Tab

Line Enable: YES

Dial Plan: (S0<:@gw0>)

PSTN Line Tab

Line Enable: Yes

VoIP-to-PSTN Gateway Enable: Yes

One Stage Dialing: Yes

Line 1 VoIP Caller DP: NONE

VoIP Answer Delay: 0

Line In Use Voltage: xx

The Line In Use Voltage needs to be lower than the Talk Battery voltage supplied by the Viking Pl-1.  Usually it is set about half way between the on-hook and off-hook voltage level.  Measure the FXO (PSTN Line) on-hook voltage by reading it on the SPA3102 INFO Tab.  Set the Line In Use Voltage substantially below the on-hook talk voltage.

If you lift the phone with the above settings you should be connected to the paging system.

Interface to an asterisk pbx system:

PSTN Line Tab

Setup Registration on PSTN Line Tab to Asterisk system. 

Register: Yes

Proxy: xxx

UserID: xxx

Password: xxx

VoIP-to-PSTN Gateway Enable: Yes

VoIP Caller Auth Method: None

One Stage Dialing: Yes

VoIP Caller Default DP: NONE

VoIP Answer Delay: 0

Line In Use Voltage: same comments as above

If you call the extension on the asterisk PBX you should be attached to the Viking unit.

Thank you very much for your assistance! Advice worked perfectly without any issue.

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