Howto:Analog Trunk (FXO) with Linksys SPA3102: Difference between revisions
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'''Innovaphone Compatibility Test Report''' | '''Innovaphone Compatibility Test Report''' | ||
{{Template:3rd Party Input}} | |||
== Linksys (Sipura) SPA:3102 == | == Linksys (Sipura) SPA:3102 == | ||
Infomation: | Infomation: | ||
*Software Version 5.1.7 (GW) | *Software Version 5.1.7 (GW) | ||
*Hardware Version 1.4.5 (a) | *Hardware Version 1.4.5 (a) | ||
Line 21: | Line 20: | ||
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. | ||
You created one SIP Trunk Without Registration between Innovaphone Gateway and Linksys SPA3102. | |||
Calls are made and received with Routes. | |||
Calls are made and | |||
===Linksys configuration=== | ===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 | |||
[[Image:Analog Trunk (FXO) with Linksys SPA3102 Linksys1.jpg]] | |||
====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 | 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 | VoIP-To-PSTN Gateway Enable:Yes | ||
PSTN Caller Auth Method:None | PSTN Caller Auth Method:None | ||
- PSTN-To-VoIP Gateway Setup | 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-To-VoIP Gateway Enable:Yes | ||
PSTN Caller Default DP:1 | PSTN Caller Default DP:1 | ||
PSTN Caller Auth Method:None | PSTN Caller Auth Method:None | ||
PSTN Caller ID Pattern:* | 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) | - FXO Timer Values (sec) | ||
PSTN Line: FXO Timer: PSTN Answer Delay: 0 | |||
PSTN | |||
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]] | |||
===innovaphone configuration=== | ===innovaphone configuration=== | ||
Configure a | 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. | |||
== Supported Codecs == | == Supported Codecs == | ||
Line 290: | Line 349: | ||
|---- | |---- | ||
|CGPN is displayed correctly | |CGPN is displayed correctly | ||
| | |yes | ||
|---- | |---- | ||
|CGPN can be supressed | |CGPN can be supressed | ||
Line 296: | Line 355: | ||
|} | |} | ||
== Related Articles == | |||
[[Howto:Cisco_Small_Business_Pro_SPA3102-EU_-_3rd_Party_Product|Cisco Small Business Pro SPA3102-EU - 3rd Party Product]] | |||
[[ | |||
Latest revision as of 09:55, 24 July 2015
Innovaphone Compatibility Test Report
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 |