Debug VoIP call pass-through¶
1. Check that incoming VoIP call is detected by the system using web interface.
For this enter Call detail report for suspected peer and search for the call.
If call is found - go to i.9.
2. On this stage we know that CDR is absent in suspected peer.
Search for CDR on all peers.
For this use Originators -> Call detail report and turn on group by name option.
Perform the same on gateways and Users list.
Likely, the call entered in a different peer.
If call is found, then incoming call is associated not with the peer you suspect.
Perform п.1 for new peer. Check system configuration.
3. On this stage we know that call is absent in system's CDR.
Check if the IP address of peer has been blocked by the Fail2Ban application.
You can check it by looking for peer's IP address in Network -> Firewall tables -> SIP deny -> Hosts.
4. Check if you don't hit the issue described in Registering on Smartswitch several SIP accounts from same IP address+port.
5. Enable Pcap capturing on originator.
After making a call, search the call in Pcap report on expected peer.
If not found - search on all peers in Pcap report, using a filter by B-number.
Perhaps, there is another peer with the same authorization settings and call is recorded on behalf of him instead of the expected peer (similar as was done in i.2).
In case if call is present on this report, but is absent in Call detail report - the call has came but the system hasn't authorized it.
Possible reasons - Registration on Smartswitch of several SIP accounts from the same IP address+port, mismatch of originator settings and Smartswitch settings regarding authentication (login mismatch, password mismatch, etc).
You can check which login sends client and from which IP address by opening pcap file in Wireshark.
In case if the call is absent in Call detail report - likely call doesn't come to Smartswitch at all.
You can re-check this with manual pcap capturing which is described in section Debug VoIP call with Wireshark.
In case you don't see a call when manually capturing pcap - you need to check settings on the originator's side.
For example, check if the originator is not blocked by his ISP, check his firewall, etc.
6. On this stage we know that call comes ingress to Smartswitch.
Check, that call handling starts.
For this use instruction Debug call handling.
In case if you see in log file call handling execution in response to incoming call - association with peer is found and call handling is started.
CDR should be present in the system.
Therefore, i.1 and i.2 have been performed incorrectly.
Repeat them.
7. On this stage we know, that VoIP call arrives, but call handling is not started.
Possible reasons: a mistake of definition of branch entering inside Call handler or mistake of peer definition in the system.
Turn on logging of debug 10 and verbose 10 using instruction Debug Asterisk.
For SIP additionally turn on debuging using instruction Debug SIP protocol in Asterisk.
In log file you'll get search for output lines after INVITE from originator.
They will contain information of why Asterisk hasn't found association with peer and hasn't started execution of Call handler.
Usually the reason is mismatch of IP addresses and authentication parameters between ones sent by originator and ones configured on Smartswitch.
8. Check that call handler is executed according to expected logic.
For this use instruction Debug call handling.
9. In case if call handling is performed correctly and Softswitch application is invoked,
however outbound call is not generated to a terminator, then Check routing with the same parameters as the Softswitch is executed in the call handling log.
In case if during Checking routing you immediately see an error message - then this error is in orignator settings.
In case if during Checking routing you see an originator price but you don't see expected routes, enter in field outbound peer the name of expected outbound peer and system will show you why there is no route to it.
10. In case if route checking shows properly and the route is to a dynamic peer (which doesn't have static IP address and is expected to register) -
check if the dynamic peer is registered on Smartswitch.
11. Enable following log levels in System -> Logging -> asterisk application: info, error, debug.
Check log file /var/log/smartswitch/asterisk.log.
See instruction How to work with log file.
Find the time of a call and its B-number and see log messages related to routing a call - it could hint you about the reason of the issue.
12. Check, if the concurrent limit or attempts per second in configured on a supplier.
When these restrictions apply, the outbound call won't be generated.
Also check all other non-default limitations configured on a supplier.