Howto:Analog Trunk (FXO) with Linksys SPA3102: Difference between revisions

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VoIP-To-PSTN Gateway Enable:Yes
VoIP-To-PSTN Gateway Enable:Yes
Line 1 VoIP Caller DP:None


PSTN Caller Auth Method:None
PSTN Caller Auth Method:None


One Stage Dialing:YES
One Stage Dialing:YES
VoIP Caller Default DP:None





Revision as of 16:27, 30 January 2009

Innovaphone Compatibility Test Report

3rd party input
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-> PSTN Line


- 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


Analog Trunk (FXO) with Linksys SPA3102 Linksys1.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.

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: *

Analog Trunk (FXO) with Linksys SPA3102 Linksys3.jpg


- VoIP-To-PSTN Gateway Setup:

VoIP-To-PSTN Gateway Enable:Yes

PSTN Caller Auth Method:None

One Stage Dialing:YES


- PSTN-To-VoIP Gateway Setup:

PSTN-To-VoIP Gateway Enable:Yes

PSTN Ring Thru Line 1:No

PSTN CID For VoIP CID:Yes

PSTN Caller Default DP:1

PSTN Caller Auth Method:None

PSTN Caller ID Pattern:*

 put  a "*" in PSTN Caller ID Pattern to enable Caller ID presentation , so calls don't show anonymous


- FXO Timer Values (sec)

PSTN Line: FXO Timer: PSTN Answer Delay: 0

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

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


Analog Trunk (FXO) with Linksys SPA3102 Linksys4.jpg

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


Analog Trunk (FXO) with Linksys SPA3102 04.png


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

Incoming calls from Linksys will come with number defined in Dialing Plan 1 (126 in the example). 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