innovaphone wiki - User contributions [en]

Howto:ONI H.323 Provider Compatibility Test

2009-08-17T11:46:07Z

Ebc suporte:

{{Template:3rd Party Input}}

== Summary ==

'''SIP Provider: ONI'''

The provider uses a SONUS system and it's working with SIP protocol. Provides IP Voice Trunks via closed IP Circuits (MLPS Network).

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF7 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-07-04T21:18:27Z

Ebc suporte: /* Broadcast Group & Waiting Queue */

{{Template:3rd Party Input}}

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Voice Trunks via closed IP Circuits (MLPS Network).

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!Tested feature
!Result
|----
|'''Caller can make a call to a Broadcast Group'''
|'''No'''
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Yes'''
|----
|'''Announcement if nobody picks up the call'''
|'''Yes'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-06-17T08:48:46Z

Ebc suporte: /* Summary */

{{Template:3rd Party Input}}

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Voice Trunks via closed IP Circuits (MLPS Network).

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-06-12T15:04:50Z

Ebc suporte: /* Summary */

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Voice Trunks via closed IP Circuits with dynamic E1 Redundancy (Automatic DDI Translation to E1 Connection if IP connectivity is lost) .

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-06-12T15:03:24Z

Ebc suporte: /* Summary */

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Voice Trunks via closed IP Circuits with E1 Redundancy.

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-06-12T14:59:53Z

Ebc suporte:

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Trunks via closed IP Circuits with E1 Redundancy.

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Testing Environment ==

[[Image:HFO_SIP_Compatibility_Test_5.PNG]]

This scenario describes a setup where the PBX and phones are in a private network. No RTP Proxy is used. Private networks are configured for correct media relay.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:ONI H.323 Provider Compatibility Test

2009-06-12T14:55:33Z

Ebc suporte: New page: == Summary == '''H.323 Provider: ONI''' The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Trunks via closed IP Circuits with E1 Redundancy. * '''Provid...

== Summary ==

'''H.323 Provider: ONI'''

The provider uses a SONUS system and it's working with H.323 protocol. Provides IP Trunks via closed IP Circuits with E1 Redundancy.

* '''Provider Homepage: http://www.oni.pt/'''

== Current test state ==

Testing of this product are in progress. Initiated at 1 June of 2009.

== Test Results ==

Bold lines in the test results indicate a KO-criteria.

=== Basic Call ===

{| border="1"
!Tested feature
!Result
|----
|'''call using g711a'''
|'''Yes'''
|----
|'''call using g711u'''
|'''Yes'''
|----
|call using g723
|No
|----
|call using g729
|Yes
|----
|Overlapped sending
|No
|----
|Fax using T.38
|No
|----
|CGPN can be supressed
|No
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|'''Voice Quality OK?'''
|'''Yes'''
|}

=== Direct Dial In ===

{| border="1"
!Tested feature
!Result
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Yes'''
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Yes'''
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|'''DTMF tones sent correctly'''
|'''Yes'''
|----
|'''DTMF tones received correctly'''
|'''Yes'''
|}

=== Hold/Retrieve ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be put on hold'''
|'''Yes'''
|----
|Held end hears music on hold / announcement from PBX
|Yes
|----
|Held end hears music on hold / announcement from provider
|Yes
|}

=== Transfer with consultation ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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) ===

{| border="1"
!Tested feature
!Result
|----
|'''Call can be transfered'''
|'''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 ===

{| border="1"
!Tested feature
!Result
|----
|Call can be transfered
|No
|----
|Held end hears dialing tone
|Yes
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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'''
|}

== Configuration ==

===Firmware version===

All innovaphone devices use V7 HF5 as firmware.

Howto:Analog Trunk (FXO) with Linksys SPA3102

2009-01-15T11:28:13Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

{{Template:3rd Party Input}}

== 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:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[image: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:YES

VoIP Caller Default DP:None

[[Image: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:*

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

- FXO Timer Values (sec)

VoIP Answer Delay:0

PSTN Answer Delay:0

[[Image: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

[[Image: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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|yes
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-28T14:48:52Z

Ebc suporte: /* Calling Party Number */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[image: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

[[Image: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

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_03.png]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_04.png]]

For last 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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|yes
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

File:Analog Trunk (FXO) with Linksys SPA3102 04.png

2008-11-28T14:46:50Z

Ebc suporte:

File:Analog Trunk (FXO) with Linksys SPA3102 03.png

2008-11-28T14:46:31Z

Ebc suporte:

File:Analog Trunk (FXO) with Linksys SPA3102 02.png

2008-11-28T14:44:41Z

Ebc suporte:

File:Analog Trunk (FXO) with Linksys SPA3102 01.png

2008-11-28T14:43:07Z

Ebc suporte:

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-28T14:40:46Z

Ebc suporte: /* innovaphone configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[image: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

[[Image: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

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_03.png]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_04.png]]

For last 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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-28T14:36:51Z

Ebc suporte: /* innovaphone configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[image: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

[[Image: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

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_03.png]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_04.png.jpg]]

For last 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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-28T14:36:20Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[image: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

[[Image: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

[[Image:Analog_Trunk_(FXO)_with_Linksys_SPA3102_03.png]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:SPA3102_d.jpg]]

For last 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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:55:40Z

Ebc suporte: /* innovaphone configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[Image:SPA3102_a.jpg]]

- 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

[[Image:SPA3102_b.jpg]]

- 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

[[Image:SPA3102_c.jpg]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:SPA3102_d.jpg]]

For last 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 ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:52:05Z

Ebc suporte: /* innovaphone configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[Image:SPA3102_a.jpg]]

- 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

[[Image:SPA3102_b.jpg]]

- 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

[[Image:SPA3102_c.jpg]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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

Create new GW Trunk.

Protocol:SIP

Mode: Gateway without Registration

Primary SIP Server: IP address of Linksys

[[Image:SPA3102_d.jpg]]

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:48:51Z

Ebc suporte: /* configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1

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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[Image:SPA3102_a.jpg]]

- 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

[[Image:SPA3102_b.jpg]]

- 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

[[Image:SPA3102_c.jpg]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:48:34Z

Ebc suporte: /* Linksys (Sipura) SPA:3102 */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

V1

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[Image:SPA3102_a.jpg]]

- 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

[[Image:SPA3102_b.jpg]]

- 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

[[Image:SPA3102_c.jpg]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:48:10Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

[[Image:SPA3102_a.jpg]]

- 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

[[Image:SPA3102_b.jpg]]

- 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

[[Image:SPA3102_c.jpg]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:46:25Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

- 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

- 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

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:45:20Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

- 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

- 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

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:44:57Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

- 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

- 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

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:44:06Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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

Menu: Voice-> PSTN Line

- Proxy and Registration:

Register:No

Make Call Without Reg:Yes

Ans Call Without Reg:Yes

- 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

- 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

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:42:10Z

Ebc suporte: /* Linksys configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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'''
'''Menu: Voice-> PSTN Line'''

- Proxy and Registration:
Register:No
Make Call Without Reg:Yes
Ans Call Without Reg:Yes

- 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

- 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

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T17:39:40Z

Ebc suporte: /* configuration */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

V1
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===

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'''
'''Menu: Voice-> PSTN Line'''

'''Proxy and Registration:'''
Register:No
Make Call Without Reg:Yes
Ans Call Without Reg:Yes

'''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.''

'''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

'''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

'''FXO Timer Values (sec)'''

VoIP Answer Delay:0
PSTN Answer Delay:0

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]

Howto:Analog Trunk (FXO) with Linksys SPA3102

2008-11-25T16:44:20Z

Ebc suporte: /* Linksys (Sipura) SPA:3102 */

'''Innovaphone Compatibility Test Report'''

== Linksys (Sipura) SPA:3102 ==

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

Information:
*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.

For connecting the Linksys for Analog Trunk ( FXO ) connection to the innovaphone Gateway/pbx you need a Gatekeeper/Registrar license.

===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]]

===innovaphone configuration===

Configure a registrar where the Linksys can register (as seen in picture below)

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.

== Supported Codecs ==

{| border="1"
!Codec
!Applies
|----
|G711
|yes
|----
|G729
|yes
|----
|G723
|yes
|----
|G726
|yes
|----
|GSM
|no
|----
|T.38 UDP
|no
|----
|G722
|No
|}

== Test Results ==

=== Basic Call ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Inbound(Sipura -> innovaphone)
|yes
|----
|Outbound(Innovaphone -> Sipura)
|yes
|}

=== DTMF ===

{| border="1"
!Tested feature
!Result
|----
|DTMF tones sent correctly
|yes
|----
|DTMF tones received correctly (audible)
|yes
|}

=== Hold/Retrieve ===

{| border="1"
!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===

{| border="1"
!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)===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|Device can transfer call
|yes
|----
|Held end hears dialing tone
|no - hears nothing
|}

=== Broadcast Group & Waiting Queue ===

{| border="1"
!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 ===

{| border="1"
!Tested feature
!Result
|----
|CGPN is displayed correctly
|no
|----
|CGPN can be supressed
|yes
|}

[[Category:AdminLog|{{PAGENAME}}]]
[[Category:RecProd Gateways|{{PAGENAME}}]]