OldTesting


Service Provider: Junction Networks
Software Revision: Asterisk 1.6.2
Micro PBX Hardware: Shuttle K4500-RS
VoIP phones: Grandstream GXP-2000 ver 1.2.2.19



1.0 Call Flow Requirements

Call Flow Requirements
Requirement ID
TC Name
Level
Result
Description
Notes
1.1.1
PBX L1 calls PBX L2
Required
PASS
PBX Line 1 Phone calls PBX Line 2 PhoneNone
1.1.2
PBX L1 calls CAS L1
Required
PASS
PBX Line 1 Phone calls PSTN Line 1None
1.1.3
CAS L1 calls PBX L1
Required
PASS
PSTN Line 1calls PBX Line 1 Phone None
1.1.4
PBX L1 calls SIP
Required
PASS
PBX Line 1 Phone calls SIP PhoneNone
1.1.5
SIP calls PBX L1
Required
PASS
SIP Phone calls PBX LineNone
1.2.1
PBX L1 calls PBX L2 Hold
Required
PASS
PBX Line 1 Phone calls PBX Line 2 Phone and L1 put L2 on/off holdNone
1.2.2
PBX L1 calls PBX L2 Hold
Required
PASS
PBX Line 1 Phone calls PBX Line 2 Phone and L2 put L2 on/off holdNone
1.2.3
PBX L1 calls CAS L1 Hold
Required
PASS
PBX Line 1 Phone calls PSTN Line 1 and PBX puts PSTN on/off holdNone
1.2.4
CAS L1 calls PBX L1 Hold
Required
PASS
PSTN Line 1calls PBX Line 1 Phone and PBX put PSTN on/off holdNone
1.2.5
PBX L1 calls SIP Hold
Required
PASS
PBX Line 1 Phone calls SIP Line 1 and PBX puts SIP on/off holdNone
1.2.6
PBX L1 calls SIP Hold
Required
PASS
PBX Line 1 Phone calls SIP Line 1 and SIP puts PBX on/off holdNone
1.2.7
SIP calls PBX L1 Hold
Required
PASS
SIP 1calls PBX Line 1 Phone and SIP puts PBX on/off holdNone
1.2.8
SIP calls PBX L1 Hold
Required
PASS
SIP 1calls PBX Line 1 Phone and PBX puts SIP on/off holdNone
1.3.1
CAS L1 calls PBX L1 attended xtfer to PBX L2
Required
PASS
CAS L1 calls PBX L1 attended xtfer to PBX L2None
1.3.2
CAS L1 call PBX L1attended xtfer to CAS L1
Required
PASS
CAS L1 call PBX L1 attended xtfer to CAS L1None
1.3.3
CAS L1 calls PBX L1attended xtfer to SIP L1
Required
PASS
CAS L1 calls PBX L1 attended xtfer to SIP L1None
1.3.4
SIP L1 calls PBX L1attended xtfer to PBX L2
Required
PASS
SIP L1 calls PBX L1 attended xtfer to PBX L2None
1.3.5
SIP L1 call PBX L1attended xtfer to CAS L1
Required
PASS
SIP L1 call PBX L1 attended xtfer to SIP L1None
1.3.6
PBX L1 calls PBX L2 and PBX L1 attended xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L1 attended xtfers to CAS L1None
1.3.7
PBX L1 calls PBX L2 and PBX L2 attended xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 attended xtfers to CAS L1None
1.3.8
PBX L1 calls PBX L2 and PBX L1 attended xtfers to SIP L1
Required
PASS
PBX L1 calls PBX L2 and PBX L1 attended xtfers to SIP L1None
1.3.9
PBX L1 calls PBX L2 and PBX L2 attended xtfers to SIP L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 attended xtfers to SIP L1None
1.3.10
PBX L1 calls CAS L1 and attended xtfers to CAS L2
Required
PASS
PBX L1 calls CAS L1 and PBX L1 attended xtfers to CAS L2None
1.3.11
PBX L1 calls CAS L1 and attended xtfers to SIP L1
Required
PASS
PBX L1 calls CAS L1 and PBX L1 attended xtfers to SIP L1None
1.3.12
PBX L1 calls SIP L1 and attended xtfers to CAS L1
Required
PASS
PBX L1 calls SIP L1 and PBX L1 attended xtfers to CAS L1None
1.3.13
SIP calls PBX L1 and SIP L1 attended xtfers to PBX L2
Required
PASS
SIP calls PBX L1 and PBX L1 attended xtfers to PBX L2None
1.3.14
CAS L1 calls PBX L1 semi-attended xtfer to PBX L2
Required
PASS
CAS L1 calls PBX L1 semi-attended xtfer to PBX L2None
1.3.15
CAS L1 call PBX L1semi-attended xtfer to CAS L1
Required
PASS
CAS L1 call PBX L1 semi-attended xtfer to CAS L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.3.16
CAS L1 calls PBX L1semi-attended xtfer to SIP L1
Required
PASS
CAS L1 calls PBX L1 semi-attended xtfer to SIP L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.3.17
SIP L1 calls PBX L1semi-attended xtfer to PBX L2
Required
PASS
SIP L1 calls PBX L1 semi-attended xtfer to PBX L2Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.3.18
SIP L1 call PBX L1semi-attended xtfer to CAS L1
Required
PASS
SIP L1 call PBX L1 semi-attended xtfer to SIP L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.1
PBX L1 calls PBX L2 and PBX L1 semi-attended xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L1 semi-attended xtfers to CAS L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.2
PBX L1 calls PBX L2 and PBX L2 semi-attended xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 semi-attended xtfers to CAS L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.4
PBX L1 calls PBX L2 and PBX L2 semi-attended xtfers to SIP L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 semi-attended xtfers to SIP L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.5
PBX L1 calls CAS L1 and semi-attended xtfers to CAS L2
Required
PASS
PBX L1 calls CAS L1 and PBX L1 semi-attended xtfers to CAS L2Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.6
PBX L1 calls CAS L1 and semi-attended xtfers to SIP L1
Required
PASS
PBX L1 calls CAS L1 and PBX L1 semi-attended xtfers to SIP L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.7
PBX L1 calls SIP L1 and semi-attended xtfers to CAS L1
Required
PASS
PBX L1 calls SIP L1 and PBX L1 semi-attended xtfers to CAS L1Observed:PBX L1 gets busy signal. Expect: PBX L1 hangs-up / disconnects.
1.4.8
SIP calls PBX L1 and SIP L1 semi-attended xtfers to PBX L2
Required
PASS
SIP calls PBX L1 and PBX L1 semi-attended xtfers to PBX L2None
1.5.1
CAS L1 calls PBX L1 blind xtfer to PBX L2
Required
PASS
CAS L1 calls PBX L1 blind xtfer to PBX L2None
1.5.2
CAS L1 call PBX L1blind xtfer to CAS L1
Required
PASS
CAS L1 call PBX L1 blind xtfer to CAS L2None
1.5.3
CAS L1 calls PBX L1blind xtfer to SIP L1
Required
PASS
CAS L1 calls PBX L1 blind xtfer to SIP L1None
1.5.4
SIP L1 calls PBX L1blind xtfer to PBX L2
Required
PASS
SIP L1 calls PBX L1 blind xtfer to PBX L2None
1.5.5
SIP L1 calls PBX L1blind xtfer to PBX L2
Required
PASS
SIP L1 calls PBX L1 blind xtfer to PBX L2None
1.5.6
PBX L1 calls PBX L2 and PBX L1 blind xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L1 blind xtfers to CAS L1None
1.5.7
PBX L1 calls PBX L2 and PBX L2 blind xtfers to CAS L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 blind xtfers to CAS L1None
1.5.8
PBX L1 calls PBX L2 and PBX L1 blind xtfers to SIP L1
Required
PASS
PBX L1 calls PBX L2 and PBX L1 blind xtfers to SIP L1None
1.5.9
PBX L1 calls PBX L2 and PBX L2 blind xtfers to SIP L1
Required
PASS
PBX L1 calls PBX L2 and PBX L2 blind xtfers to SIP L1None
1.5.10
PBX L1 calls CAS L1 and blind xtfers to CAS L2
Required
PASS
PBX L1 calls CAS L1 and PBX L1 blind xtfers to CAS L2None
1.5.11
PBX L1 calls CAS L1 and blind xtfers to SIP L1
Required
PASS
PBX L1 calls CAS L1 and PBX L1 blind xtfers to SIP L1None
1.5.12
PBX L1 calls SIP L1 and blind xtfers to CAS L1
Required
PASS
PBX L1 calls SIP L1 and PBX L1 blind xtfers to CAS L1None
1.6.1
Verify the PBX L1conference with PBX L2 and CAS L1
Required
PASS
Verify the PBX L1conference with PBX L2 and CAS L1None
1.6.2
Verify the PBX L1conference with CAS L2 and CAS L1
Required
PASS
Verify the PBX L1conference with CAS L2 and CAS L1None
1.6.3
Verify the PBX L1conference with CAS L1 and SIP L1
Required
PASS
Verify the PBX L1conference with CAS L1 and SIP L1None
1.7.1
Call Waiting
Required
NT
Verify the ability of the uPBX to support call waiting for multiple inbound calls on one line.Not Tested (NT) because the uPBX is configured to ring an alternate line on the VoIP phone for inbound calls if the first line is busy.
ID
TCNAME
Required
RESULT
DescriptionNone


2.0 Interoperability Requirements

2.1 Junction Networks Interoperability

2.1 Junction Networks Interoperability
Requirement ID
TC Name
Level
Result
Description
Notes
2.1.1
Registration
Required
PASS
The uPBX_v1 shall register with Junction NetworksNone
2.1.2
Registration
Required
PASS
The standard approach is used to challenge all requests (the Proxy-to-User Authentication scheme as outlined in Section 22.3 of RFC 3261). Customers are identified by their username. If identification by username fails, the authorization username is used. None
2.1.3
Caller-ID
Required
FAIL
If the username is a valid phone number (ten or eleven digits), it is used as the caller-id number associated with a call. Otherwise, if the display name is a valid phone number, it is used as the caller-id number associated with a call.Observed: Calling name delivery in inbound INVITE to external VoipVoip SIP phone. Expected: CNAME delivery.
2.1.4
iLBC Support
Optional
N/A
Check if your VoIP device supports iLBC compressed codec and is chosen as default or preferred codec. iLBC codec works better if you have very low speed of Internet connection or high latency such as satellite Internet connection. You can find codec options usually in advanced configuration of your VoIP device.None
2.1.5
Long Call
Required
RESULT
Verify uPBX_v1 is capable of sustaining a call for over 2 hours.None
2.1.6
Numbering
Required
PASS
The standard e.164 numbering plan (ITU) is used. North American numbers are required to be prefixed with a '1'.None
2.1.7
Incoming DID
Required
PASS
Verify incoming calls based on the DID that was called, registering with Junction Networks can be helpful if a customer subscribes to more than one DID. This server is recommended for use with SIP servers and IP PBXes.Verified only 1 DID.
2.1.8
NAT Traversal
Required
PASS
Use SIP keep-alive packets or set your Registration interval below the NAT binding expiration time (90 seconds will usually suffice).Verified Junction Networks sets its Expires header to 120 seconds. Verified Registration requests are being sent out at 105 second intervals.
2.1.9
REFER
Optional
N/A
Verify the uPBX supports REFER requests as defined in RFC 3515 .Verified the IP PBX does not use REFER for call transfers.
2.1.10
Comfort Noise
Optional
PASS
Verify the uPBX supports comfort noise as specified in RFC 3389.Verified the IP PBX does not use comfort noise but transfers the held party to MOH (Music On Hold).
2.1.11
Silence Suppression
Required
PASS
Verify Voice Activity Detection (VAD) can be disabled.Verfied the IP PBX does not use Silence Suppression.
2.1.12
911 or E911 Support
Optional
FAIL
Verify Junction Networks supports 911 or E911.Verified that Junction Networks does not support 911 or E911.
2.1.13
DID Routing
Optional
NT
DID Routing provides optional extended service features such as contingency failover, routing calls to static IP addresses, routing calls utilizing multiple protocols, routing calls back into the PSTN, and adjustable timeouts.Not Tested.
2.1.14
Stun Server Support
Optional
N/A
Verify Junction Networks Stun Server support.Not needed with Junction Networks.
2.1.15
Failover
Optional
NT
Failover Scenarios: 1.) Your device is not registered at all. There is therefore nowhere to send the INVITE. We fail over immediately. 2.) 5 SIP or IAX INVITE requests - 1 per second for 5 seconds - without a return response. Fail-over after 5 seconds. 3.) Rejected INVITE. If you send back "congestion" or otherwise respond to the INVITE but respond in the negative, we fail over immediately upon receipt of the negative response. 4.) If you have a SIP device and it does not pick up the call after x seconds (the seconds are up to you with a default of 120), we fail over after x seconds of ring-no-answer.Not Tested.
2.1.16
Variable Length RFC2833
Optional
FAIL
Verify the Junction Networkssupports variable length DTMF transmission using RFC2833. Verify that a digit press over 2 seconds sends 2 seconds of RFC2833 RTP Events.Verified that Junction Networks sends a fixed RFC2833 digit on-time of 800ms.
2.1.17
Minimum DTMF Cycle Time
Required
PASS
Verify the Junction Networks sends DTMF inband/outband with a cycle time of no less than 93 ms.Verified Junction Networks cycle time greater than 93ms.
2.1.18
Minimum Inter-Digit Time
Required
PASS
Verify a minimum interdigit time of no less than 40 ms.Verified interdigit time greater than 200ms.
2.1.19
Minimum Digit On-Time
Required
PASS
Verify a minimum DTMF digit on-time of no less than 40 ms.Verified Junction Networks sends a fixed RFC2833 digit on-time of 800ms.
ID
TCNAME
Required
RESULT
DescriptionVerified Junction Networks only sends 800ms of DTMF as RFC2833.


3.0 Functional Requirements

3.1 Signaling and Routing

3.1 Signaling and Routing
Requirement ID
TC Name
Level
Result
Description
Notes
3.1.1
RFC 3261
Required
PASS
Verify the uPBX call processing with Junction Networks using SIP RFC 3261None
3.1.2
UDP Transport for SIP
Required
PASS
Verify the uPBX will use UDP to transport SIP signaling. None
3.1.3
Symetric Signaling
Required
PASS
Verify the uPBX supports sending and receiving on the same UDP port.None
3.1.4
Defined UDP ports
Required
PASS
Verify the uPBX supports configurable UDP SIP signaling ports 5060 and above.Verified the port can be set in /etc/asterisk/sip.conf with the parameter bindport=5060. However, I did not test this functionality.
3.1.5.1
P-Assert-ID Support
Optional
NT
Verify the uPBX support of the PAI headerPAI is not supported as an option but can be added in the extensions.conf as a SipAddHeader() function.
3.1.5.2
Remote Part ID Support
Optional
PASS
Verify the uPBX support of the RPID headerVerified the uPBX supports RPID.
3.1.6
Diversion Support
Optional
PASS
Verify the uPBX support of the Diversion header.Verified the IP PBX used a Diversion header in a 302 Moved Temporarily from a phone. However, the call-forwarding did not work to the PSTN user.
3.1.7
E.164 Support
Optional
N/A
Verify the uPBX supports E.164 numbering format in the SIP URL.Not a requirement for Junction Networks.
3.1.8
Host Field
Required
PASS
Verify the uPBX supports IPv4 or FQDN in the SIP URL.None
3.1.9
Tel URL
Optional
PASS
Verify the uPBX ability to enable/disable the user=phone in the SIP URL.None
3.1.10
Full and Compact Header
Required
PASS
Verify the uPBX ability to support full and compact headers.Verified the uPBX could be configured to use Compact headers but did not verify functionality:Compact SIP headers: No in sip.conf
3.1.11
DTMF in INFO
Required
PASS
Verify the uPBX does not use the SIP INFO method for sending DTMF.None
3.1.12
Call Hold
Required
PASS
The uPBX_v1 shall support Call Hold using either the RFC3264 method (a=sendonly) or RFC2543 method (c=0.0.0.0). It is preferred that the RFC3264 method be used.Verified the IP PBX uses RFC3264 a=sendrec and redirects the audio to MOH.
3.1.13
Early offer
Required
PASS
Verify the uPBX support of Early Offer using SIP 183 Session Progress Message with SDP.Verified the uPBX sends DTMF events to the Service Provider upon receipt of a 183 Session Progress with SDP.
3.1.14
Receiving 503
Optional
NT
After receiving a 503 Service Unavailable response, The uPBX_v1 shall attempt to forward the request to an alternate server. Not Tested. Created a Tracker for a uPBX enhancement.
3.1.15
Sending 503
Optional
FAIL
The uPBX_v1 shall only send a 503 Service Unavailable if: 1) it is temporarily unable to process the call and 2) the customer’s network architecture includes an alternate element that may be able to successfully process the same call.Verified the uPBX does not send ANY messages out when unable to process a call.
3.1.16
Receiving 604
Required
NT
Verify the uPBX does not try a second proxy with receipt of a 6xx message.Not Tested.
3.1.17
Sending 604
Required
FAIL
The uPBX_v1 shall send a 6XX response to Junction only when indicating that the uPBX is not able to fulfill a request.Verified the uPBX does not send anything out when unable to process a call. Created a Tracker for a uPBX enhancement.
3.1.18
302 Support
Required
PASS
Verify the uPBX does not send a 302 Moved Temporarily to Junction Networks server.Verified the uPBX does not send a 302 when a PHONE attempts call forwarding. Did not see a way to easily configure a user in the uPBX to call forward.
3.1.19.1
Privacy Header Support
Required
FAIL
Verify the uPBX supports sending the Privacy header as defined in RFC 3323. Verified the uPBX does not use the Privacy Header for CallerID blocking. Opened a Tracker for a uPBX enhancement.
3.1.19.2
Privacy Header Support
Required
NT
Verify the uPBX supports receiving the Privacy header as defined in RFC 3323. Did not have a phone that set privacy.
3.1.20
Receipt of Diversion Header
Required
PASS
Verify the uPBX supports the receipt of the Diversion header from a call forwarded phone and properly populates the Caller ID fields.Verified the uPBX supports receipt of the Diversion header and properly populates the Caller ID field with the originating caller and no the forwarded phone.
3.1.21
DNS SRV and A Records
Optional
PASS
Verify the uPBX supports DNS SRV and A record queries and responses.Verified the uPBX supports SRV queries and A - Record responses.
3.1.22
DNS TTL
Optional
PASS
Verify the uPBX refreshes the DNS A records after the TTL has expired.Verified the uPBX refreshes the DNS records upon every outbound call.
3.1.23
DNS Failover
Optional
NT
Verify the uPBX supports multiple DNS entries and tries each entry in the event the previous DNS entries are unavailable.Not Tested
3.1.24
DNS Proxy failover
Optional
RESULT
Verify the uPBX will try each proxy in the list of proxies returned in an SRV response in the event the primary proxy(s) are unavailable.None
3.1.25
DNS Priorities
Optional
RESULT
Verify the uPBX obeys the priorities returned in the SRV response.None
3.1.26
DNS Weights
Optional
RESULT
Verify the uPBX obeys the weights returned in the SRV response.None
3.1.27
DNS Ports
Optional
RESULT
Verify the uPBX uses the Ports returned in the SRV response.None
3.1.28
DNS Unsupported
Optional
RESULT
Verify if the uPBX does not support DNS services that the uPBX supports a primary and secondary proxies using IPv4 addresses.None
3.1.29
NAT Support
Required
PASS
Verify the uPBX callp and voice path NAT traversal support.Verified the uPBX traverses a NAT.
3.1.30
Session Timer Support
Optional
RESULT
Verify the ability of the uPBX to support Session Timers and the ability to update the session time using either the UPDATE or re-INVITE methods. None
3.1.31
Delayed SDP Support
Required
RESULT
Verify the uPBX callp and path using INVITEs without SDP (Delayed SDP).None
3.1.32
SIP-T Support
Optional
RESULT
Verify the uPBX callp and path using SIP-T.None
3.1.33
Unknown and Proprietary Headers
Required
RESULT
Verify the uPBX ignores any unknown or proprietary headers not associated to the uPBX.None
3.1.34
SDP Codec Offer
Required
PASS
Verify the uPBX SDP offers with a list of codecs in order of preference.Verified the uPBX only offer 0 101 for G.711 codec and RFC2833.
3.1.35
SDP Codec Answer
Optional
PASS
Verify the uPBX SDP answers with only one codec from the list offered.Verified the uPBX only responded with 0 101 for G.711 and RFC2833.
3.1.36.1
RFC2833 DTMF
Required
PASS
Verify the uPBX supports the reception and transmission of RFC 2833 DTMF. Verified the uPBX negotiated RFC2833. Also, verified the Grandstream GXP-2000 ver 1.2.2.19 transits and receives RFC2833 RTP events.
3.1.36.2
RFC2833 DTMF
Required
PASS
Verify the uPBX FXS port supports the reception and transmission of RFC 2833 DTMF. Verified the IP PBX transmission and reception of RFC2833 DTMF events to the VoIP Phone Adapter.
3.1.37
In-band DTMF
Required
RESULT
Verify the uPBX support the reception and transmission of in-band DTMF.None
3.1.38.1
Variable Length RFC2833
Optional
PASS
Verify the uPBX supports variable length DTMF transmission using RFC2833. Verify that a digit press over 2 seconds sends 2 seconds of RFC2833 RTP Events.Verified the uPBX supports RFC2833 from the IP Phone for voicemail.
3.1.38.2
Variable Length RFC2833
Optional
PASS
Verify the uPBX FXS supports variable length DTMF transmission and receiption using RFC2833. Verify that a digit press over 2 seconds sends 2 seconds of RFC2833 RTP Events.None
3.1.39
Minimum DTMF Cycle Time
Required
PASS
Verify the uPBX sends DTMF inband/outband with a cycle time of no less than 93 ms.Verified the uPBX IP phone send RFC2833 greater than 93ms.
3.1.40
Maximum Inter-Digit Time
Required
PASS
Verify the uPBX supports DTMF RTP Events with gaps (interdigit time) of up to 2 seconds between key presses.Verified variable length DTMF.
3.1.41
Minimum Inter-Digit Time
Required
PASS
Verify a minimum interdigit time of no less than 40 ms.Verified minimum interdigit time of 76ms.
3.1.42
Minimum Digit On-Time
Required
PASS
Verify a minimum DTMF digit on-time of no less than 40 ms.Using Wireshark verified minimum digit on-time 640ms.
3.1.43
Comfort Noise
Required
PASS
Verify the uPBX supports the transmission of comfort noise and the ability to disable silence suppression (no VAD) Verified Silence suppression is disable and the uPBX uses MoH.
3.1.44
RTCP Support
Optional
RESULT
Verify the uPBX supports RTCP statistics transmission and reception.None
3.1.45
RTP Port Addressing
Required
PASS
Verify the uPBX supports a configurable RTP port and address range.Verified the uPBX IP Phones support configurable RTP ports or dynamic RTP ports and ranges.
3.1.46
Signaling DiffServ AF32
Required
PASS
Verify the ability of the uPBX to classify SIP signaling traffic with a DiffServ DSCP encoding of AF32 (Assured Forwarding Class 3 Medium Drop) per RFC2597.Verified the uPBX can be configured to use AF32 for SIP signaling.
3.1.47
Media DiffServ EF
Required
PASS
Verify the ability of the uPBX to classify RTP Media traffic with a DiffServ DSCP encoding of EF (Expidited Forwarding) per RFC3246Verified the uPBX can be configured to use EF for RTP.
3.1.48
T.38 Support
Optional
NT
Verify the ability of the uPBX to support the T.38 Fax protocol. Verify that T.38 can be re-negotiated to G.711 if the endpoint does not support T.38.No test equipment to test this feature.
3.1.49
2100Hz Detection
Optional
FAIL
Verify the ability of the uPBX to detect 2100Hz fax tone and that it upspeeds to G.711 if using a compression codec, disables NLP (Non Linear Processing and that the echo cancelers are enabled at the endpoint.Verified the IP PBX leaves the echo cans enable while faxing. Verified only a 50% successful fax rate with FXS card.
3.1.50.1
FXS Fax Performance of 95%
Required
FAIL
Verify the ability of the uPBX FXS to send and receive faxes with a greater than 95% success rate.Verified the uPBX only has a 50% successful fax rate with a 12 page fax using the FXS port.
3.1.50.2
ATA Fax Performance of 95%
Required
PASS
Verify the ability of the uPBX ATA to send and receive faxes with a greater than 95% success rate.Verified the uPBX ATA can transmit and receive a 12 page fax with over a 95% success rate.
3.1.51
IPSec Support
Optional
RESULT
Verify the ability of the uPBX to support IPSec tunnels to service providers that require IPSec Authentication.None
3.1.52
Signaling Security
Optional
RESULT
Verify the ability of the uPBX to support application level authentication and encryption.None
3.1.52
Media Security
Optional
RESULT
Verify the ability of the uPBX to support media encryption.None
3.1.53
N11 Services
Required
PASS
Verify the ability of the uPBX to support N11 calling for supported N11 numbers.Verified uPBX Callp only for N11 services.
3.1.54
Operator Services
Required
PASS
Verify the ability of the uPBX to support 0 and 00 dialing for Operator Services.Verified uPBX callp only.
3.1.55.1
E911 Emergency Services
Required
PASS
Verify the ability of the uPBX to dial 911 for Emergency Services if the Service Provider supports E911.Verified uPBX callp dialing N11 Services using the VoIP trunk.
3.1.55.2
911 Emergency Services
Required
PASS
Verify the ability of the uPBX to dial 911 for Emergency Services on the local PSTN line.None.
3.1.56
Toll Free Services
Required
PASS
Verify the ability of the uPBX to support 1-800 and other toll-free originations.Verifed the ability of the uPBX callp and path dialing 1800 numbers.
3.1.57
Dialed Number Blocking
Required
PASS
Verify the ability of the uPBX to block certain number like International dialing, 976 and 1-900 numbers.Verified the uPBX 976 numbers could be blocked with a fake service provider provisioned and 1900 number can be blocked by not adding routing in the Calling Rules table.
3.1.58
Voice Quality G.711 Codec
Required
PASS
Verify the uPBX supports a minimum voice quality PESQ score of 4.0 for the G.711 codec.See results below in section 5.0.
3.1.59
Voice Quality G.729 Codec
Required
NT
Verify the uPBX supports a minimum voice quality PESQ score of 3.0 for G.729a codecs and 3.5 for G.729 native codecs.Not Tested.
3.1.60
Early Media Support
Optional
PASS
Verify the ability of the uPBX to support early media transmission with reception of a 183 Session Progress with SDP.Verified the uPBX supports the reception of the 183 Session Progress and the transmission of RTP events with early media. Did not verify voice path.
3.1.61
REFER Call Forwarding
Optional
N/A
Verify the ability of the uPBX to support call forwarding by REFER method.Verified the uPBX uses the INVITE method for call fowarding.
3.1.62
Diversion Call Forwarding
Required
RESULT
Verify the ability of the uPBX to support call forwarding using the Diversion header.None
3.1.63
OPTION Method Support
Required
NT
Verify the ability of the uPBX to respond to a received OPTION message.Not tested.
3.1.64
G.711 Support
Required
PASS
Verify the ability of the uPBX to support G.711 payload type 0 with a 20ms packetization.Verified the uPBX callp and path using G.711 mulaw codec.
3.1.65
G.729 Support
Required
NT
Verify the ability of the uPBX to support G.729 payload type 18 with a 20ms packetization.Not Tested.
3.1.66
Called Number Length
Required
PASS
Verify the ability of the uPBX to support 3, 7, 10, and 11 dialed digit lengths.Verified the uPBX callp dialing 3,7,10 and 11 digits.
3.1.67
E.164 Support
Optional
NT
Verify the ability of the uPBX to support + dialing in the E.164 addressing format.E.164 support was not tested.
3.1.68
Privacy Header Support
Required
RESULT
Verify the ability of the uPBX to send the Privacy Header to the PSTN if set.None
3.1.69
CNAME Support
Required
RESULT
Verify the ability of the uPBX to support the reception of the CNAME and populate the display field.None
3.1.70
International Dialing
Required
PaSS
Verify the ability of the uPBX to support International dialing.Verified the ability of the uPBX to dial the International access code 011.
ID
TCNAME
Required
RESULT
DescriptionNone
ID
TCNAME
Required
RESULT
DescriptionNone


4.0 Security Requirements

4.1 Security Requirements

4.1 uPBX Security Requirements
Requirement ID
TC Name
Level
Result
Description
Notes
ID
SSH Support
Required
RESULT
Verify the uPBX supports SSH.None
ID
TCNAME
Required
RESULT
DescriptionNone
ID
TCNAME
Required
RESULT
DescriptionNone


5.0 Voice Quality Results

5.1 uPBX Voice Quality Results
ServiceProvider
Side
MOS
Noise
Gain
Codec
Delay
PSTN Baseline
PSTN
4.26
13 dBrnC
-16 dB
PCM
2.1 ms
Junction Networks
PSTN
3.91
15 dBrnC
-12 dB
G.711
433.3 ms
Junction Networks
uPBX
4.0
14 dBrnC
-20 dB
G.711
436.9 ms
VoipVoip
uPBX
3.90
11 dBrnC
-20 dB
G.711
283.3 ms
VoipVoip
PSTN
3.95
13 dBrnC
-19 dB
G.711
283.3 ms

See RFC 2119 for definitions of the IETF Requirement Levels
http://www.ietf.org/rfc/rfc2119.txt (external link)

6.0 uPBX Features

6.1 uPBX GUI

6.1 GUI
Requirement ID
TC Name
Level
Result
Description
Notes
6.1.1
Add Users
Required
PASS
Verify the ability to add a uPBX user.
Verified the ability to add a user for a SIP phone and analog FXS port.
6.1.2
Conferencing
Required
RESULT
DescriptionNone
6.1.3
Voicemail
Required
RESULT
DescriptionNone
6.1.4
Call Queues
Required
RESULT
DescriptionNone
6.1.5
Add Service Providers
Required
PASS
Verify the ability to add a Service Provider.Verified the ability to add a Service Provider.
6.1.6
Setup Hardware
Required
RESULT
DescriptionNone
6.1.7
mIDSN Config
Required
RESULT
DescriptionNone
6.1.8
Add Calling Rules
Required
PASS
Verify the ability to add a calling rule.Verified the ability to add a calling rule and change a calling rule.
6.1.9
Incoming Calls
Required
RESULT
DescriptionNone
6.1.10
Add Voice Menus
Required
FAIL
Verify the ability to add a Voice Menu message.Verified the ability to add a Voice Menu but found that the GUI corrupted the exentsions.conf file.
6.1.11
Delete Voice Menus
Required
PASS
Verify the ability to delete a Voice Menu message.Verified the ability to delete a Voice Menu message.
6.1.12
Call Parking
Required
RESULT
DescriptionNone
6.1.13
Ring Groups
Required
PASS
Verify the functionality of Ring Groups.Verified uPBX provisioning and functionality using Ring Groups.
6.1.14
Record a Menu
Required
PASS
Verify the ability to add a Voice Menu message.Verified the ability to add a Voice Menu and play back the voice menu using the GUI.
6.1.15
Backup
Optional
FAIL
Verify the Backup Feature in the uPBX GUIScreen failed to load when Backup was selected.
6.1.16
Options
Required
RESULT
DescriptionNone


6.2 General Features

6.2 General Features
Requirement ID
TC Name
Level
Result
Description
Notes
6.2.1
Automated Attendant
Optional
RESULT
Verify the Automated Attendant feature.None
6.2.2
Call Detail Records
Optional
RESULT
Verify the ability to collect Call Detail Records.None
6.2.3
Call Forward
Required
RESULT
Verify the ability to Call Forward.None
6.2.4
Call Monitoring
Optional
RESULT
Verify the ability to monitor calls.None
6.2.5
Call Parking
Optional
RESULT
Verify the ability to park calls.None
6.2.6
Call Queuing
Optional
RESULT
Verify the ability to queue calls.None
6.2.7
Call Recording
Optional
RESULT
Verify the ability to record calls.None
6.2.8
Call Transfer
Required
RESULT
Verify the ability to transfer calls.None
6.2.9
Call Waiting
Required
RESULT
Verify the call waiting ability.None
6.2.10
Caller ID
Required
RESULT
Verify the ability of the uPBX to receive caller-id, send caller-id and pass-through caller-id.None
6.2.11
Do Not Disturb
Optional
RESULT
Verify the ability to se DND.None
6.2.12
E911
Required
RESULT
Verify the ability to dial E911.None
6.2.13
Music On Hold
Optional
RESULT
Verify MoH with hold callsNone
6.2.14
Music On Transfer
Optional
RESULT
Verify MoH with call transfers.None
6.2.15
Privacy
Optional
RESULT
Verify the ability of the uPBX to set Privacy and pass-through privacy calls.None
6.2.16
Three-way Calling
Required
RESULT
Verify the ability of the uPBX to support three-way calling.None
6.2.17
Voicemail
Required
RESULT
Verify the ability of the uPBX to keep and store voicemail.None
6.2.18
Web Configuration
Optional
RESULT
Verify the ability to configure the uPBX with a web browser GUI.None
6.2.19
Options
Required
RESULT
DescriptionNone





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