Hello new here I have Inbound Route/ and SIP Trunk that receives several DID and they are routing to our main IVR of the company but some call are getting “The number you have call is out of Service” ( Intermittent ) below is the configuration of the Inbound route and SIP Trunk
Description: Main Corp IVR
DID Number: 1XXXXXXXXXX
CID Priority Route:
CID name prefix:
Music On Hold: PM-BILAT
Pause Before Answer:
Privacy Manager: No
IVR : Main I VR
Trunk Name: NAME
Outbound CallerID: blank
CID Options: Allow my CID
Maximum Channels: Blank
Disable Trunk: Uncheck
Monitor Trunk Failures: Blank
Dialed Number Manipulation Digits BLANK
Trunk Name NAME
Your trunking provider may be sending calls from more than one IP address. On a failing call, your log likely contains “Rejecting unknown SIP connection from …” or "Received incoming SIP connection from unknown peer”. If so, find out what addresses your provider can send calls from. If there are only two or three, define a trunk for each additional address. If there are many, see https://community.asterisk.org/t/received-incoming-sip-connection-from-unknown-peer-from-registered-sip-provider/69721 for one approach, or consider using a pjsip trunk, where you can simply list the addresses or networks in the Match field.
If the above is not your issue or you still have trouble, post a log of a failing call.
Stewart thank you the response I already did work session with DID provider and they sent PCAP trace showing FreePBX answering the call and IP that is coming from is allowed. Like I said I am new at FreePbx I tried to capture the call on the Logs that portal shows me but was not able to there is a section where I can send command can i use this to capture a failing call
Reports -> Asterisk Logfiles by default will show you 500 lines, which should be plenty. Make a call to the PBX that fails, then click Show. If you have trouble with that, just search the file /var/log/asterisk/full for the relevant info.
Please confirm that the call was sent from the exact address specified by the host=x.x.x.x for your trunk. If not, what do you mean by ‘allowed’? (With default settings) if Asterisk gets a call from an IP address that does not match a trunk, it will route to a ‘not in service’ message.
Did the number in the SIP URI (what’s after ‘INVITE sip:’ in the pcap they sent) match an Incoming Route? Try defining a catch-all route (DID number left blank) and see whether the problem calls are caught by that.