Howto:AEMCOM SIP Provider Compatibility Test

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Innovaphone Compatibility Test Report

Contents

Summary

SIP Provider: AEMCOM


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

AEMCOM does not support the remote hold feature. When making a blind transfer(using redial key) the remote end will not get a MOH/dialtone from the provider.

AEMCOM has no T.38 capability however we successfully tested sending and receiving faxes over g711.

AEMCOM has achieved 86% of all possible test points. For more information on the test rating, please refer to Test Description


  • Features:
    • Direct Dial In
    • DTMF
  • Supported Codecs by the provider
    • G711a/u

Current test state

The tests for this product have been completed.

Testing of this product has been finalized July 07th, 2009.

Testing Enviroment

Scenario NAT

Image:HFO_SIP_Compatibility_Test_5.PNG

This scenario describes a setup where the PBX and phones are in a private network. No stun server was required during while testing. 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 No
call using g729 No
Overlapped sending No
early media channel Yes
Fax using T.38 No
CGPN can be supressed No
Long time call possilbe (>30 min) 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 dialing tone No

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: 7.00 hotfix7 IP800[09-70300.17]
  • IP24: 7.00 hotfix5 IP24[09-70300.14]
  • IP200A: 7.00 hotfix5 IP200A[09-70300.14]
  • IP230: 7.00 hotfix5 IP230[09-70300.14]
  • IP230: 7.00 hotfix7 IP230[09-70300.17]

SIP - Trunk

First of all the SIP Trunk must be configured. Here an example of our AEMCOM - Trunk.

Image:AEMCOM SIP Compatibility Test - Trunk.jpg

AEMCOM awaits in the From Header the complete Calling Party Number(CGPN). The default innovaphone setting is to not send the complete CGPN in the FROM - Header, but in the Preffered Identity Header. Change the setting From Header: to CGPN in user part of URI.

Image:AEMCOM SIP Compatibility Test - Trunk detail.jpg

Number Mapping

The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.

Image:AEMCOM SIP Compatibility Test - Mapping.jpg

Route Settings

Because AEMCOM, 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 suplementary 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:AEMCOM SIP Compatibility Test - Routes.jpg

Now the PBX and the phones are setup correctly. You should be able to make call in both directions.

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