Howto:Colt VoIP Access SIP Provider Compatibility Test: Difference between revisions

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=== Firmware version ===
=== Firmware version ===


* version 9.00 hotfix3
* version 9.00 hotfix4
 
=== SIP - Trunk ===
 
First of all the SIP Trunk must be configured. Since Colt authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX. Make sure that Media Relay is activated and an exclusive codec is selected.
 
[[Image:Colt VoIP Access SIP Provider Compatibility Test 1.PNG]]
 
=== Number Mapping ===
 
[[Image:Colt VoIP Access SIP Provider Compatibility Test 2.PNG]]
 
=== Route Settings ===
 
Because Colt, 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 your 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.
 
It is also very important that the correct number is sent to the SIP provider. To ensure this the gateway 'Routing Table' must contain a Clip No Screening configuration, as shown below.
 
[[Image:Colt VoIP Access SIP Provider Compatibility Test 3.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. That's why the T.38 codec must be enabled. Additionally an exclusive codec must be selected at the analog interface.
 
[[Image:Colt VoIP Access SIP Provider Compatibility Test 4.PNG]]
 
===CLIR===
 
To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser:
 
<nowiki>http://PBX-IP-address/!config add SIP /pai</nowiki>
<nowiki>http://PBX-IP-address/!config write</nowiki>
<nowiki>http://PBX-IP-address/!config activate</nowiki>


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

Revision as of 11:10, 19 October 2011

Innovaphone Compatibility Test Report

Summary

SIP Provider: Colt

  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711A
    • G729
    • T.38

Current test state

This product is being tested right now. The test is not yet completed.


Testing Environment

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 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, requires Media Relay & STUN
CGPN can be suppressed Yes
CLIP no screening Yes
Reverse Media Negotiaton Yes
Long time call possible Yes
External Transfer Yes, requires Media Relay
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 sent correctly via SIP-Info 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

Transfer with consultation

Tested feature Result
Call can be transferred 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 transferred Yes
Held end hears music on hold or dialling tone Yes
Call returns to transferring device if the third

Endpoint is not available

Yes

Blind Transfer

Tested feature Result
Call can be transferred Yes
Held end hears dialling tone Yes

Broadcast Group & Waiting Queue

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

Configuration

Firmware version

  • version 9.00 hotfix4

SIP - Trunk

First of all the SIP Trunk must be configured. Since Colt authenticates a user account only by IP - Address, you cannot use the normal SIP-Gateway object. You need to configure a GW without registration and route the calls to the PBX. Make sure that Media Relay is activated and an exclusive codec is selected.

Colt VoIP Access SIP Provider Compatibility Test 1.PNG

Number Mapping

Colt VoIP Access SIP Provider Compatibility Test 2.PNG

Route Settings

Because Colt, 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 your 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.

It is also very important that the correct number is sent to the SIP provider. To ensure this the gateway 'Routing Table' must contain a Clip No Screening configuration, as shown below.

Colt VoIP Access SIP Provider Compatibility Test 3.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. That's why the T.38 codec must be enabled. Additionally an exclusive codec must be selected at the analog interface.

Colt VoIP Access SIP Provider Compatibility Test 4.PNG

CLIR

To suppress the CGPN for outbound calls a config line option must be activated. Please enter the following lines in your browser:

http://PBX-IP-address/!config add SIP /pai
http://PBX-IP-address/!config write
http://PBX-IP-address/!config activate