Howto:Tele2 (Austria) SIP Provider Compatibility Test: Difference between revisions

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(New page: '''Innovaphone Compatibility Test Report''' -- {{Template:Compat Status "in progress"}} -- == Summary == '''SIP Provider: Tele2(Austria)''' * Features: ** Direct Dial In ** Fax over T...)
 
 
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'''Innovaphone Compatibility Test Report'''
'''Innovaphone Compatibility Test Report'''
-- {{Template:Compat Status "in progress"}} --


== Summary ==
== Summary ==
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** G729
** G729
** T.38
** T.38
'''SIP Provider: Tele2 (Austria)'''
The provider supports all required innovaphone features and is therefore qualified as [[Howto:What_is_a_%22recommended_product%22%3F#SIP_Provider|recommended SIP Provider]].
Tele2(Austria) is only available in Austria. They use a complete different backbone infrastructure then [[Howto:Tele2(Sweden) SIP Provider Compatibility Test |Tele2(Sweden)]]
That beeing said, Tele2 (Austria) has achieved 96% of all possible test points. For more information on the test rating, please refer to [[Howto:SIP_Interop_Test_Description#Summary|Test Description]]






== Current test state ==
== Current test state ==
{{Template:Compat Status "tested"(sip provider)}}
{{Template:Compat Status "tested"}}
<!-- {{Template:Compat Status "in progress"}} -->
<!-- {{Template:Compat Status "in progress"}} -->
<!--{{Template:Compat Status "certified"|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}}-->
<!--{{Template:Compat Status "certified"|certificate=Tpl_sip.business_Toplink_SIP_Provider_-_product-cert.pdf}}-->
<!-- {{Template:Compat Status "rejected"}} -->
<!-- {{Template:Compat Status "rejected"}} -->


<!-- Testing of this product has been finalized October 23th, 2007.-->
Testing of this product has been finalized July 7th, 2008.


== Testing Enviroment ==
== Testing Enviroment ==


=== Scenario ===
=== Scenario NAT ===
This scenario tries to be as simple as possible. The IP - PBX and the phones are all in the same network, having public IP - adresses.
 
The signalling channel will pass through the PBX, the media channel will go from endpoint (phone) to endpoint (provider).
[[Image:HFO_SIP_Compatibility_Test_5.PNG]]
 
This scenario describes a setup where the PBX and phones are in a private network. The IP800 must use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.


[[Image:CompatProviderSIP-1.JPG]]


== Test Results ==
== Test Results ==
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|----
|----
|'''call using g711u'''
|'''call using g711u'''
|''''''
|'''Yes'''
|----
|----
|call using g723
|call using g723
|
|Yes
|----
|----
|call using g729
|call using g729
|
|'''Yes'''
|----
|----
|Overlapped sending
|Overlapped sending
|
|No
|----
|----
|'''early media channel'''
|'''early media channel'''
|''''''
|'''Yes'''
|----
|----
|Fax using T.38
|Fax using T.38
|
|Yes
|----
|'''Long time call possible(>30 min)'''
|'''Yes'''
|----
|----
|CGPN can be supressed
|CGPN can be supressed
|
|Yes (only extension can be surpressed)
|----
|----
|'''Reverse Media Negotiaton'''
|'''Reverse Media Negotiaton'''
|''''''
|'''Yes'''
|----
|----
|'''Voice Quality OK?'''
|'''Voice Quality OK?'''
|''''''
|'''Yes'''
|}
|}


Line 85: Line 98:
|----
|----
|'''Inbound(Provider -> Innovaphone)'''
|'''Inbound(Provider -> Innovaphone)'''
|''''''
|'''Yes'''
|----
|----
|'''Outbound(Innovaphone -> Provider)'''
|'''Outbound(Innovaphone -> Provider)'''
|''''''
|'''Yes'''
|}
|}


Line 98: Line 111:
|----
|----
|'''DTMF tones sent correctly'''
|'''DTMF tones sent correctly'''
|''''''
|'''Yes'''
|----
|----
|'''DTMF tones received correctly'''
|'''DTMF tones received correctly'''
|''''''
|'''Yes'''
|}
|}


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|----
|----
|'''Call can be put on hold'''  
|'''Call can be put on hold'''  
|''''''
|'''Yes'''
|----
|----
|Held end hears music on hold / announcement from PBX
|Held end hears music on hold / announcement from PBX
|
|Yes
|----
|----
|Held end hears music on hold / announcement from provider
|Held end hears music on hold / announcement from provider
|
|No
|}
|}


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|----
|----
|'''Call can be transfered'''
|'''Call can be transfered'''
|''''''
|'''Yes'''
|----
|----
|Held end hears music on hold
|Held end hears music on hold
|
|Yes
|----
|----
|'''Call returns to transferring device if the third  
|'''Call returns to transferring device if the third  
Endpoint is not available'''
Endpoint is not available'''
|''''''
|'''Yes'''
|}
|}


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|----
|----
|'''Call can be transfered'''
|'''Call can be transfered'''
|''''''
|'''Yes'''
|----
|----
|Held end hears music on hold or dialing tone
|Held end hears music on hold or dialing tone
|
|Yes
|----
|----
|'''Call returns to transferring device if the third  
|'''Call returns to transferring device if the third  
Endpoint is not available'''
Endpoint is not available'''
|''''''
|'''Yes'''
|}
|}


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|----
|----
|Call can be transfered
|Call can be transfered
|
|Yes
|----
|----
|Held end hears dialing tone
|Held end hears music on hold
|
|Yes
|}
|}


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|----
|----
|'''Caller can make a call to a Broadcast Group'''
|'''Caller can make a call to a Broadcast Group'''
|''''''
|'''Yes'''
|----
|----
|'''Caller can make a call to a Waiting Queue'''
|'''Caller can make a call to a Waiting Queue'''
|''''''
|'''Yes'''
|----
|----
|'''Announcement if nobody picks up the call'''
|'''Announcement if nobody picks up the call'''
|''''''
|'''Yes'''
|}
|}


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*IP800: 6.00 sr1-hotfix6 IP800[07-60600.77]
*IP800: 6.00 sr2-hotfix6 IP800[07-60900.77]
*IP22: 6.00 sr1-hotfix6 IP22[07-60600.77]
*IP22: 6.00 sr2-hotfix6 IP22[07-60900.77]
*IP200: 6.00 sr1-hotfix6 IP230[07-60600.77]
*IP200: 6.00 sr2-hotfix6 IP200[07-60900.77]
*IP230: 6.00 sr1-hotfix6 IP230[07-60600.77]
*IP230: 6.00 sr2-hotfix6 IP230[07-60900.77]


=== SIP - Trunk ===
=== SIP - Trunk ===
Set an SIP Trunk via Administration/Gateway/Interface/Sip
Important is to set the right port (5082), and a stun server if the setup is behind a Nat router
[[image:Howto-Tele2_(Austria)_SIP_Provider_Compatibility_Test_Tele2_sip.png]]
It is important that on the Trunk Object (in the Pbx ) the flag Outgoing Calls no Name is set.
That means that in the P-Preferred-Identity only the extension will be sent (and so the cgpn will work correctly)
[[image:Tele2_(Austria)_SIP_Provider_Compatibility_Test_Tele2_trunk.PNG]]




=== Number Mapping ===
=== Number Mapping ===
There is no need to map any cgpn numbers as Tele 2 Austria need s only the extension Number and adds the access number(Trunk number) automatically.
Also for incoming calls the cdpn don't need to be mapped as Tele2 (Austria) sends only the extension to the innovaphone pbx.




=== Route Settings ===
=== Route Settings ===
Because Tele2 (Austria), as most SIP - Providers too, doesn't support overlap sending, you must enable the blockwise sending of the phone number. You can do this by enabling Force enblock in the automatically generated Routes.
The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this checkbox is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.
[[image:Howto-Tele2_(Austria)_SIP_Provider_Compatibility_Test_Tele2_1.png]]




=== Fax ===
=== Fax ===


Now the PBX and the phones are setup correctly. You should be able to make call in both directions and send and receive fax messages.
 
 
The FAX was connected via a IP22 to the IP800. When configuring the IP22 you must keep in mind that you will use the analog interface for fax communication. Thats why the T.38 codec must be enabled.
 
[[Image:Tele2 SIP Compatibility Test 5.PNG]]
 
 
 
 
Now the PBX is setup correctly. You should be able to make call in both directions and send and receive fax messages.




[[Category:Compat|{{PAGENAME}}]]
[[Category:Compat|{{PAGENAME}}]]

Latest revision as of 17:51, 3 March 2010

Innovaphone Compatibility Test Report

Summary

SIP Provider: Tele2(Austria)

  • Features:
    • Direct Dial In
    • Fax over T.38
    • DTMF
  • Supported Codecs by the provider
    • G711
    • G729
    • T.38


SIP Provider: Tele2 (Austria)

The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.

Tele2(Austria) is only available in Austria. They use a complete different backbone infrastructure then Tele2(Sweden)

That beeing said, Tele2 (Austria) has achieved 96% of all possible test points. For more information on the test rating, please refer to Test Description



Current test state

The tests for this product have been completed.

Testing of this product has been finalized July 7th, 2008.

Testing Enviroment

Scenario NAT

HFO SIP Compatibility Test 5.PNG

This scenario describes a setup where the PBX and phones are in a private network. The IP800 must use a stun server, in order to send correct SIP - messages. The IP800 works as media relay, all RTP - streams go through the PBX.


Test Results

For more information on the test procedure, please read the following wiki article: SIP Interop Test Description. Bold lines in the test results indicate a KO-criteria.


Basic Call

Tested feature Result
call using g711a Yes
call using g711u Yes
call using g723 Yes
call using g729 Yes
Overlapped sending No
early media channel Yes
Fax using T.38 Yes
Long time call possible(>30 min) Yes
CGPN can be supressed Yes (only extension can be surpressed)
Reverse Media Negotiaton Yes
Voice Quality OK? Yes

Direct Dial In

Tested feature Result
Inbound(Provider -> Innovaphone) Yes
Outbound(Innovaphone -> Provider) Yes

DTMF

Tested feature Result
DTMF tones sent correctly Yes
DTMF tones received correctly Yes

Hold/Retrieve

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 No

Transfer with consultation

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)

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

Tested feature Result
Call can be transfered Yes
Held end hears music on hold Yes

Broadcast Group & Waiting Queue

Tested feature Result
Caller can make a call to a Broadcast Group Yes
Caller can make a call to a Waiting Queue Yes
Announcement if nobody picks up the call Yes

Configuration

General Information

Firmware version


  • IP800: 6.00 sr2-hotfix6 IP800[07-60900.77]
  • IP22: 6.00 sr2-hotfix6 IP22[07-60900.77]
  • IP200: 6.00 sr2-hotfix6 IP200[07-60900.77]
  • IP230: 6.00 sr2-hotfix6 IP230[07-60900.77]

SIP - Trunk

Set an SIP Trunk via Administration/Gateway/Interface/Sip

Important is to set the right port (5082), and a stun server if the setup is behind a Nat router


Howto-Tele2 (Austria) SIP Provider Compatibility Test Tele2 sip.png


It is important that on the Trunk Object (in the Pbx ) the flag Outgoing Calls no Name is set.

That means that in the P-Preferred-Identity only the extension will be sent (and so the cgpn will work correctly)

Tele2 (Austria) SIP Provider Compatibility Test Tele2 trunk.PNG


Number Mapping

There is no need to map any cgpn numbers as Tele 2 Austria need s only the extension Number and adds the access number(Trunk number) automatically.

Also for incoming calls the cdpn don't need to be mapped as Tele2 (Austria) sends only the extension to the innovaphone pbx.


Route Settings

Because Tele2 (Austria), as most SIP - Providers too, doesn't support overlap sending, you must enable the blockwise sending of the phone number. You can do this by enabling Force enblock in the automatically generated Routes.

The second setting you must check is Interworking(QSIG,SIP). This feature must be enabled to properly relay supplementary services, like Hold over the SIP Trunk. If this checkbox is unchecked only basic call Information like connect and disconnect will be forwarded by the Gateway.


Howto-Tele2 (Austria) SIP Provider Compatibility Test Tele2 1.png


Fax

The FAX was connected via a IP22 to the IP800. When configuring the IP22 you must keep in mind that you will use the analog interface for fax communication. Thats why the T.38 codec must be enabled.

Tele2 SIP Compatibility Test 5.PNG



Now the PBX is setup correctly. You should be able to make call in both directions and send and receive fax messages.