I am looking for help to solve the problem that we are facing in our Phone Server with version FreePBX 14.0.3.6.
Everything seems to be working fine but when we are able to call out side using our trunk its working fine but at that time we will not able to receive call from outside. But sometime when we are able to receive a call at that time we will not able to make call outside using our trunk so either one works at a time.
Do you know how your system is connected to the PSTN? We don’t, and the answer is critical to being able to troubleshoot. Logs from a call (out of /var/log/asterisk/full, for example) would also be helpful.
I keep on getting those error which is unknown extension also i can make outgoing calls without any problem but incoming calls work sometimes but not all the times and FreePBX configured through microtic router so please guide me or help to solve this problem.
@Auroraj I’m not sure how “core show calls” or “core show channels” is going to help with this.
If you are having issues making an outbound call while another call is active what is the actual response to the attempted outbound call? Do you get a busy signal? Does it ring? Are you getting some sort of “Number can’t be dialed” error playback?
If you are having issues with inbound calls, but only sometimes, is there a second call happening during this? Are there no calls happening?
We need actual call traces/sip debugs not looking at how many active calls or channels are happening on the system. We need to see the actual INVITE transactions for these calls to see if the PROVIDER is not letting outbound calls go through. We need to see if the inbound calls are actually hitting the system.
Outbound and inbound calling are two different things. Outbound doesn’t require the call to go through your firewall/NAT as new connection. If they are denying outbound calls at some point, the debug will provide the reasons why. Inbound calls are subject to your firewall/NAT rules and can cause issues with inbound calls not making it to the PBX at all.
So you need to try and replicate these issues so you can get the proper call traces/sip debugs. To get those you need to do:
asterisk -rvvvvvvvvvvv
sip set debug on
Get the ENTIRE call that fails and post it here (or put it on pastebin and post that link)