Howto:Analog Trunk (FXO) with Linksys SPA3102

Innovaphone Compatibility Test Report

this is 3rd party content not provided by innovaphone, see history for authors.

Linksys (Sipura) SPA:3102

Infomation:

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

innovaphone gateway/pbx

This information applies to

• all PBX Platforms

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

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.

Linksys configuration

Login: Admin - Advance Mode

Menu: Voice-> Line 1 and PSTN Line

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

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

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: *

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".

- 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

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

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.

Supported Codecs

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

Test Results

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

Dial Inward

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

DTMF

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

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

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

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

Blind Transfer

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

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

Calling Party Number

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

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