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

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Infomation:
Infomation:
*Software Version 3.3.6(GW)
*Hardware version 1.3.5(a)


V1


Information:
*Software Version 5.1.7 (GW)
*Software Version 5.1.7 (GW)
*Hardware Version 1.4.5 (a)
*Hardware Version 1.4.5 (a)
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With this configuration you can dial the whole number at once . you don´t have to wait for the analog dial tone.
With this configuration you can dial the whole number at once . you don´t have to wait for the analog dial tone.


For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.
You created one SIP Trunk Without Registration between Innovaphone Gateway and Linksys SPA3102.


V1
Calls are made and received with Routes.
 
In this configuration is created one SIP Trunk Without Registration between Innovaphone Gateway and Linksys SPA3102.
Calls are made and receive with Routes.


===Linksys configuration===
===Linksys configuration===
Configuration of the proxy settings
Proxy: Ip address of the innovaphone Gateway/Pbx
user id: the registration name to the innovaphone Gateway/Pbx
[[image:Analog_trunk_with_Linksys_SPA3102_Clipboard01.png]]
Voip to PSTN gateway enable set to yes
Line Voip caller DP set to none
One stage dialing set to no
[[image:Analog_trunk_with_Linksys_SPA3102_Clipboard02.png]]
V1


Login: Admin - Advance Mode
Login: Admin - Advance Mode
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===innovaphone configuration===
===innovaphone configuration===


Configure a registrar where the Linksys can register (as seen in picture below)
Configure a Gateway without registration
 
administration/gateway/voip
 
[[image:Analog_trunk_with_Linksys_SPA3102_Sipura3.png]]
 
 
Configure a route to the Linksys
 
number out is here a 0 - you can take any digit this is for the analog trunk assignment.
 
Then you configure a " ^ " this indicates an Delay for one second , then the rest of the number will be dialed in dtmf with 300msec delay between every digit.
 
[[image:Analog_trunk_with_Linksys_SPA3102_Sipura4.png]]
 
 
For incoming calls (from analog Trunk to innovaphone) you have to configure the proper routes - from the GW where the Sipura is connected to the pbx.
 
V1


Gateway->VoIP
Gateway->VoIP
Line 154: Line 108:




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

Revision as of 17:33, 28 November 2008

Innovaphone Compatibility Test Report

Linksys (Sipura) SPA:3102

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

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:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

Analog Trunk (FXO) with Linksys SPA3102 01.png


- Dial Plans:

Dial Plan 1: (S0<:126@192.168.0.254)

In the example 126 is the extension we desire to calls be redirected and 192.168.0.254 the IPBX IP Address. S0<: means dial in Linksys like a hotline.


- VoIP-To-PSTN Gateway Setup:

VoIP-To-PSTN Gateway Enable:Yes

Line 1 VoIP Caller DP:None

PSTN Caller Auth Method:None

One Stage Dialing:No

VoIP Caller Default DP:None

Analog Trunk (FXO) with Linksys SPA3102 02.png


- 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:*


- FXO Timer Values (sec)

VoIP Answer Delay:0

PSTN Answer Delay:0

Analog Trunk (FXO) with Linksys SPA3102 03.png

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
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