Debug call hangup reason¶

Perform next instructions consecutively.

1. Enter Call detail report of a call of interest.

2. In case if call is absent in detail report, the call hasn't gone through the system.
Perform debug of a call pass-through according to instruction Debug VoIP call pass-through.

3. Check who is hangup initiator in CDR.
In case if hangup initiator is peer - the call has been terminated by remote side with hangup cause, which is specified in hangup cause field.
You can check this by Debug VoIP call with Wireshark or Debug SIP protocol in Asterisk
In case if hangup initiator - system - go to i.12.

4. Check duration of call connection.
In case if call has been connected and answered duration is greater than 1-2 sec - go to i.7.

5. Check that signalling session has been correctly established between Smartswitch and remote side.
For this perform Debug VoIP call with Wireshark or Debug SIP protocol in Asterisk.
In case if you have difficulties in understanding the capture, you can provide it to Streamco technical support.

6. In case if there is an issue in signalling from Smartswitch side - appeal to Streamco technical support and provide capture, retrieved in i.5.
In case if there is an issue in signalling from the remote side, or is there are no obvious issues in signalling, go to i.11, because it's not obvious from our side why remote side has decided to terminate the call.

7. There is no issues in signalling, but possible issue with RTP stream pass-through.
Check that in tab RTP in call details there are present RTP frames, sent to remote side.
If yes - go to i.11.

8. In case if there is a linked CDR, then check than RTP stream has come from remote side in linked CDR.
If yes - then stream has been blocked on Smartswitch level.
Appeal to Streamco technical support and provide captures of both call legs.

• check Smartswitch configuration

• check addresses of media streams, which are present in signalling.
  Perform Debug VoIP call with Wireshark or Debug SIP protocol in Asterisk.
  In case if you have difficulties with understanding the capture you can provide it to Streamco technical support.

In case if RTP stream goes directly - go to i. 11.

10. RTP stream hasn't come from remote side.
Check configuration of network and firewall.
Appeal to representative of remote peer to explain why there is no RTP from him.

11. Appeal to representative of remote side for the explanation about a reason for call termination.

12. Check if there is a linked CDR.
If yes and you look at inbound leg - the call hasn't pass through the system.
Perform debug of acording to instruction Debug VoIP call pass-through.
If no and you're looking on outbound leg - the call has been generated by Smartswitch, go to i.14.

13. Check who is hangup initiator in linked CDR.
If this is - peer, then original call leg has been terminated because remote side has hanged up linked leg.
Investigate the reason of call hangup of linked leg according to this instruction.

14. Decision about call hangup has been made by Smartswitch.
This could be in one of the cases:

• the signalling negotiation hasn't finished successfully
  The remote side hasn't answered by the message which is expected on the appropriate stage of session negotiation, or hasn't answered at all.
  You can check this by Debug VoIP call with Wireshark or Debug SIP protocol in Asterisk.

• balance cut-off has been hit
  You can check this by checking peer balance.

• answered call has been hanged up due to RTP timeout on SIP channel(if there was no RTP from remote side during some configured time interval)
  You can check this by looking at RTP tab in call details in Call detail report.
  Or by Debug SIP protocol in Asterisk.

• the call has been hanged up via web-interface

• the call has been hanged up due to execution of VAS
  You can check this by Debug Asterisk.

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