Howto:QSC IPfonie extended - SIP Provider Compatibility Test
Innovaphone Compatibility Test Report
Summary
SIP Provider: QSC
The provider supports all required innovaphone features and is therefore qualified as recommended SIP Provider.
Please note that in order to use Clip no screening, some extra configuration must be done.
That being said, the provider has achieved 96.2% of all possible test points (128 on 133). For more information on the test rating, please refer to Test Description
- Features:
- T.38 IP Fax
- Direct Dial In
- DTMF
- CGPN can be suppressed
- CLIP No Screening
- SIP over TCP
- Supported Codecs by the provider
- G711a/u
- G722
- G729
- G723
- T.38 UDP
Current test state
The tests for this product have been completed and it has been approved as a recommended product (Certification document).
Testing of this product has been finalized October 11th, 2012.
Testing Enviroment
This scenario describes a setup where the PBX and phones are in a private network.
The SIP trunk is configured without Media Relay, no STUN and without exclusive coder.
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 | OK |
| call using g711u | OK |
| call using g723 | OK |
| call using g729 | OK |
| call using g722 | OK* |
| Overlapped sending | NOK |
| early media channel | OK |
| Fax using T.38 | OK |
| Reverse Media Negotiation | OK |
| CGPN can be suppressed | OK |
| CLIP no screening | OK |
| Long time call possible(>30 min) | OK |
| External Transfer | OK |
| NAT Detection | OK |
| Redundancy | OK |
| SIP over TCP | OK |
| Voice Quality OK? | OK |
"*" - G722 is only supported if the remote endpoint also supports this codec (QSC doesn't do transcoding of the calls).
Direct Dial In
| Tested feature | Result |
|---|---|
| Inbound(Provider -> Innovaphone) | OK |
| Outbound(Innovaphone -> Provider) | OK |
DTMF
| Tested feature | Result |
|---|---|
| DTMF tones sent correctly | OK |
| DTMF tones sent correctly via SIP-Info | NOK |
| DTMF tones received correctly | OK |
Hold/Retrieve
| Tested feature | Result |
|---|---|
| Call can be put on hold | OK |
| Held end hears music on hold / announcement from PBX | OK |
Transfer with consultation
| Tested feature | Result |
|---|---|
| Call can be transferred | OK |
| Held end hears music on hold | OK |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? | MoH Ok? |
|---|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | OK | OK |
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | OK | OK |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | OK | OK |
| sip-provider-phone calls inno1. inno1 transfers to inno2. | OK | OK |
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. | OK | OK |
Transfer with consultation (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transferred | OK |
| Held end hears music on hold or dialling tone | OK |
| Call returns to transferring device if the third
Endpoint is not available |
OK |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? | MoH Ok? |
|---|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | OK | OK |
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | OK | OK |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | OK | OK |
| sip-provider-phone calls inno1. inno1 transfers to inno2. | OK | OK |
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. | OK | OK |
Blind Transfer
| Tested feature | Result |
|---|---|
| Call can be transferred | OK |
| Held end hears dialling tone | OK |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? |
|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | OK |
| inno1 calls sip-provider-phone. inno1 transfers to inno2. | OK |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | OK |
| sip-provider-phone calls inno1. inno1 transfers to inno2. | OK |
| sip-provider-phone calls inno1. sip-provider-phone transfers to inno2. | OK |
Blind Transfer (alerting only)
| Tested feature | Result |
|---|---|
| Call can be transferred | OK |
| Held end hears dialling tone | OK |
The following tests are made to test if call transfer is working.
| Tested feature | Voice Ok? |
|---|---|
| inno1 calls inno2. inno2 transfers to sip-provider-phone. | OK |
| inno1 calls sip-provider-phone. sip-provider-phone transfers to inno2. | OK |
| sip-provider-phone calls inno1. inno1 transfers to inno2. | OK |
Broadcast Group & Waiting Queue
| Tested feature | Result |
|---|---|
| Caller can make a call to a Broadcast Group | OK |
| Caller can make a call to a Waiting Queue | OK |
| Announcement if nobody picks up the call | OK |
Configuration
Firmware version
All innovaphone devices use V9 hotfix 16 build 9.061101 as firmware.
SIP - Trunk
Here's the configuration of the SIP gateway interface.
Number Mapping
Route Settings
Force Enblock Setting it's required for outgoing calls. QSIG/SIP Interworking setting it's recommended.
Redundancy
- QSC allows more than one registration on the same account. Incoming calls will be delivered on the first device that registered on the account. If the first device goes down or looses its registration, then incoming calls are delivered on the 2nd registered device. When the initial device comes back online then calls will delivered again to it.
Known Issues
- In order to CLIP No Screening work properly we must set option to send SIP Address P-Asserted Identity like below.
http://x.x.x.x/!config add SIP /pai http://x.x.x.x/!config write http://x.x.x.x/!config activate