Keep Talk Telecommunication Sdn Bhd

PROBLEM TYPE: "CONNECTION"

The main objective of this problem type is to determine why:

I. Incoming calls are not working

II. Outgoing calls are not working

III. Calls are interrupted after a certain time

PROBLEM TYPE: GENERAL "CONNECTION PROBLEM" CHECKS"

Check the operational configuration in the Portal:
a. Check the PBX OrgUnit:
- Is there a blocking Public Call Permission responsible?
b. Check the PBX Dashboard:
- Is there the limit of external calls reached?
c. Check the PBX Settings:
- Is the PBX still active (date valid)

PROBLEM TYPE: "NO INCOMING CONNECTION"

Check the Trace:
Was there an incoming INVITE from the PSTN with the B number? If no INVITE was received:
- Check with the PSTN provider, why this number is not routed to this anSwitch.

PROBLEM TYPE: "NO OUTGOING CONNECTION"

Check in the Call List if the call was routed in the anSwitch to the PSTN or OnNet destination:

a. Check the dialed B number:
- Is the dialed B number correct public PSTN number?
- Is the dialed B number a OnNet public PSTN number?

b. Check the PBX Settings of A:
- Is the TopStop of the PBX reached?

c. Check in the Extension Settings of A:
- Is there a blocking Public Call Permission responsible?

d. If B is a OnNet destination, check in its Extension Features:
- Are the assigned phones registered?

e. Check the Trace:
Was there an incoming INVITE with the B number from the A user? 
If no INVITE was received:
• Check with the A user, why this call did not work:
- Phone defect?
- Phone not correct connected?
- Phone not registered  is it correct configured?
- No internet access?

•  Was there an INVITE from the anSwitch to the PSTN or OnNet destination?
- Is in this INVITE the B number still correct?
- Was there any SIP cause response from the PSTN provider like:
- 4xx: Failure responses from the B side
- 6xx: No route was found in the routing table

 If no INVITE was sent:
- Check the SIP cause response of the anSwitch
- Check the Support Log with the call-ID of this call and search for any hints

OVERVIEW "INTERRUPTED CONNECTION"

Interrupted connections can have multiple reasons:

I. Charge limitation by a TopStop

II. Unintentional "hooking on" of the phone by a user

III. Connection supervision by the phones (Session Timer)

IV. Media stream supervision by any devise in the telephone system e.g., no RTP packets during 30sec

V. Phone defect

** I. - II. are the most common reasons.

 I. Check interruption by TopStop:
a. Check the PBX Settings of A:
- Is the TopStop of the PBX reached?

II. Check for unintended "hook on" of a user:
a. Search the call in the PBX Calls list > Select Show Call Stats
b. Check in the Technical Info:
- SIP Status
- Cause of Release(Q850)
For details on the SIP response codes, see: http://en.wikipedia.org/wiki/List_of_SIP_response_codes

 III. Check the call supervision by the SIP Session Timer:
a. Search the call in the PBX Calls list > Select Show Call Stats
b. Check in the Technical Info:
- SIP Status
- Cause of Release(Q850)
c. Search the call in the PBX Calls list > Select Download SIP Trace
d. Analyze the trace for timed out "Session timers"
- "Session Timer" within the dialog is expired without renewal.
- It is typically renewed every 180 - 300sec via RE-INVITE
- The renegotiated is initiated by the pre-negotiated side (refresher)

 IV. Check for RTP media problems if the release reason of the SIP trace points to the PSTN:
a. Search the call in the PBX Calls list > Select Show Call Stats
b. Check in the Quality Info for:
- High packet loss