Howto:SilverServer SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
Summary
SIP Provider: SilverServer(Austria)
The provider does not support all required innovaphone features and is therefore not qualified as recommended SIP Provider.
For more information on the test rating, please refer to Test Description
Current test state
The tests for this product could not be completed or not all mandatory tests were passed. See the Summary section for more details.
Testing of this product has been finalized April 26th, 2011.
Testing Enviroment
Scenario
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 | No |
| call using g729 | No |
| Overlapped sending | No |
| early media channel | Yes |
| Fax using T.38 | No |
| CGPN can be supressed | No |
| Reverse Media Negotiaton | Yes |
| Voice Quality OK? | Yes |
Direct Dial In
| Tested feature | Result |
|---|---|
| Inbound(Provider -> Innovaphone) | Yes |
| Outbound(Innovaphone -> Provider) | No, CGPN not displayed correctly at PSTN phone |
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 | No |
| 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 | No |
| 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 | No |
| 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: 6.00 dvl-sr1 IP800[07-60600.74]
- IP22: 6.00 sr1-hotfix4 IP22[07-60600.72]
- IP200: 6.00 dvl-sr1 IP230[07-60600.58]
- IP230: 6.00 dvl-sr1 IP230[07-60600.58]
SIP - Trunk
First of all the SIP Trunk must be configured. Since SilverServer 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. Also SilverServer uses different servers for load balancing reasons. You must make an GW without registration object for every possible IP-destination(e.g. SilverServer SIP Server).
Like described, before you need a Gateway for every possible SilverServer SIP Server. Note that the Server Adrress on the two GWs changes, while the Domain entry does not. Another important setting is to change the SIP Interop Tweak from AOR to AOR with CGPN as display on both GWs(i.e. GW1 & GW3). You have to do this, because SilverServer does not read the Preffered identity header information but looks directly in the FROM header for DDI information.
Number Mapping
The complicated part on this issue is the correct mapping of the outgoing and incoming numbers.
Route Settings
Because SilverServer, 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 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 and send and receive fax messages.