Howto:AEMCOM SIP Provider Compatibility Test: Difference between revisions
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Revision as of 17:53, 3 March 2010
Innovaphone Compatibility Test Report
Summary
SIP Provider: AEMCOM
- Provider Homepage: http://www.aemcom.net/
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.