h1. 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.

9. Check if RTP stream has gone directly between peers.
* 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]].

[[Отладка причины завершения вызова|Русский перевод]]