Howto:Analog Trunk (FXO) with Linksys SPA3102

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Innovaphone Compatibility Test Report

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Contents

[edit] Linksys (Sipura) SPA:3102

Infomation:

  • Software Version 5.1.7 (GW)
  • Hardware Version 1.4.5 (a)

[edit] innovaphone gateway/pbx

This information applies to

  • all PBX Platforms

6.00 dvl-sr2 IP800[07-60698]or higher

[edit] configuration

With this configuration you can dial the whole number at once . you don´t have to wait for the analog dial tone.

You created one SIP Trunk Without Registration between Innovaphone Gateway and Linksys SPA3102.

Calls are made and received with Routes.

[edit] Linksys configuration

Login: Admin - Advance Mode

Menu: Voice-> Line 1 and PSTN Line

[edit] Line1

Proxy and Registration


Line1: SIP Port 5061

Line1: Proxy and Registration: Register: no

Line1: Proxy and Registration: Make Call Without Reg: yes

Line1: Proxy and Registration: Ans Call Without Reg: yes


Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys1.jpg

[edit] PSTN line

PSTN Line: SIP Port 5060

PSTN Line: Proxy and Registration: Register: no

PSTN Line: Proxy and Registration: Make Call Without Reg: yes

PSTN Line: Proxy and Registration: Ans Call Without Reg: yes


Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys2.jpg


Dial Plans


PSTN Line: Dial Plan 1: (S0<:@xx.zz.yy.ww)

In the example calls are redirected to 172.16.88.99 the IPBX IP Address. S0<: means dial in Linksys like a hotline.

VoIP-To-PSTN Gateway Setup


VoIP-To-PSTN Gateway Enable:Yes

PSTN Caller Auth Method:None

One Stage Dialing:YES

PSTN Line: Voip-To-PSTN GW: Line 1 Voip Caller DP: none

PSTN Line: Voip-To-PSTN GW: Voip Caller Default DP: none

PSTN Line: Voip-To-PSTN GW: Voip Caller ID Pattern: *

Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys3.jpg



PSTN-To-VoIP Gateway Setup

PSTN-To-VoIP Gateway Enable:Yes

PSTN Caller Default DP:1

PSTN Caller Auth Method:None

PSTN Caller ID Pattern:*

Only set a "*" in PSTN Caller ID Pattern when having problems with displaying the caller ID , otherwise leave empty. Also try to set "PSTN CID For VoIP CID" to "yes".


Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys5.jpg

- FXO Timer Values (sec)

PSTN Line: FXO Timer: PSTN Answer Delay: 0

PSTN Line: Disconnect Tone: 480@-30,620@-30;4(.25/.25/1+2)

Insert here the string for the diconnect tone of your PSTN. This can vary from country to country. If the string doesnt match the actual tone you will encounter problems when external calls are directed to a voicemail because after the caller hangs up the linksys wont disconnect.

Examples:

  • US: 480@-30,620@-30;4(.25/.25/1+2)
  • UK: 400@-30,400@-30; 2(3/0/1+2)
  • France: 440@-30,440@-30; 2(0.5/0.5/1+2)
  • Germany: 440@-30,440@-30; 2(0.5/0.5/1+2)
  • Netherlands: 425@-30,425@-30; 2(0.5/0.5/1+2)
  • Sweden: 425@-10; 10(0.25/0.25/1)
  • Norway: 425@-10; 10(0.5/0.5/1)
  • Italy: 425@-30,425@-30; 2(0.2/0.2/1+2)
  • Spain: 425@-10; 10(0.2/0.2/1,0.2/0.2/1,0.2/0.6/1)
  • Portugal: 425@-10; 10(0.5/0.5/1)
  • Poland: 425@-10; 10(0.5/0.5/1)
  • Denmark: 425@-10; 10(0.25/0.25/1)
  • New Zealand: 400@-15; 10(0.25/0.25/1)
  • Australia: 425@-13; 10(0.375/0.375/1)


PSTN Line: International Control: Line-In-Use Voltage: 15

Note the 15 Voltage setting is only necessary when connecting an IP22/IP24/IP28/IP302 for testing the analog Trunk line, because our analog line having 25 Volts on on-hook mode


Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys4.jpg

[edit] innovaphone configuration

Configure a Gateway without registration

Gateway->VoIP

Create new GW Trunk.


Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

Set the Local port to 5060


Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys6.png


Then just create routes for Incoming and Outgoing calls to Linksys Gateway Trunk created.

The route for calls to the Linksys enable enblock dialing

Incoming calls from Linksys will come with number defined in Dialling Plan 1. All Calls from Innovaphone Gateway to Linksys SPA3102 will be routed through FXO Interface directly.

Caller ID is displayed correctly when receiving calls from SPA3102.

[edit] Supported Codecs

Codec Applies
G711 yes
G729 yes
G723 yes
G726 yes
GSM no
T.38 UDP no
G722 No


[edit] Test Results

[edit] Basic Call

Tested feature Result
call using g711a yes
call using g711u yes
call using g723 yes
call using g729 yes
Overlapped sending yes
early media channel not tested
Fax not tested
Voice Quality OK? yes


[edit] Dial Inward

Tested feature Result
Inbound(Sipura -> innovaphone) yes
Outbound(Innovaphone -> Sipura) yes


[edit] DTMF

Tested feature Result
DTMF tones sent correctly yes
DTMF tones received correctly (audible) yes


[edit] Hold/Retrieve

Tested feature Result
Device can put call on hold yes
Held end hears music on hold yes
Device can terminate either call and retrieve remaining call yes


[edit] Transfer with consultation

Tested feature Result
Device can transfer call yes
Held end hears music on hold yes
Call returns to transferring device if the third

Endpoint is not available

yes


[edit] Transfer with consultation (alerting only)

Tested feature Result
Device can transfer call yes
Held end hears music on hold or dialing tone yes
Call returns to transferring device if the third

Endpoint is not available

yes


[edit] Blind Transfer

Tested feature Result
Device can transfer call yes
Held end hears dialing tone no - hears nothing


[edit] Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group yes
Caller can make a call to a Waiting Queue yes
Announcement if nobody picks up the call yes


[edit] Calling Party Number

Tested feature Result
CGPN is displayed correctly yes
CGPN can be supressed yes

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