Howto:AEMCOM SIP Provider Compatibility Test

Innovaphone Compatibility Test Report

Summary

SIP Provider: AEMCOM

• Provider Homepage: http://www.lineacom.it/

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

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.

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.

Number Mapping

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

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.

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