Howto:Tele2 (Austria) SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
-- This product is being tested right now. The test is not yet completed. --
Summary
SIP Provider: Tele2(Austria)
- Features:
- Direct Dial In
- Fax over T.38
- DTMF
- Supported Codecs by the provider
- G711
- G729
- T.38
Current test state
The tests for this product have been completed. See the Summary section for more details.
Testing Enviroment
Scenario
Scenario NAT
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 | Yes |
early media channel | No |
Fax using T.38 | 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 sr1-hotfix6 IP800[07-60600.77]
- IP22: 6.00 sr1-hotfix6 IP22[07-60600.77]
- IP200: 6.00 sr1-hotfix6 IP230[07-60600.77]
- IP230: 6.00 sr1-hotfix6 IP230[07-60600.77]
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
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)
Number Mapping
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 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.
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.