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- Jul 13, 2017
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Hi all,
Just installed the 3CX Linux iso as a VM. I've got outbound & internal calls working OK, but having problems with inbound routes. When an inbound call occurs the phone rings for a fraction of a second then hangs up.
This occurs for the main trunk line, and all other DIDs that have been setup for ring groups, individual extensions etc.
If I test ringing a single extension that has no call forwards in place...
In the 3CX admin page I see:
07/13/2017 9:26:15 AM - [CM503003]: Call(C:5): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 192.168.30.110:2051
If I tcpdump on the phone and download the PCAP, I can see that the PBX itself is sending the cancel:
Session Initiation Protocol (CANCEL)
Request-Line: CANCEL sip:[email protected]:2057;line=xoq4bnoa SIP/2.0
Method: CANCEL
Request-URI: sip:[email protected]:2057;line=xoq4bnoa
Request-URI User Part: 402
Request-URI Host Part: 192.168.30.60
Request-URI Host Port: 2057
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.30.11:5060;branch=z9hG4bK-524287-1---f2039d7c41957367;rport
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]:5060;nf=e>;tag=34ad3a54
Call-ID: yNutAUDrMPjlFsLvgufmJw..
CSeq: 1 CANCEL
Sequence Number: 1
Method: CANCEL
User-Agent: 3CXPhoneSystem 15.5.1694.0 (1694)
Reason: SIP;cause=480;text="NO_ANSWER"
Reason protocols: SIP
Content-Length: 0
So this is a test from a mobile phone to a DID that goes directly to the extension in question, which shows up as being "green" in the admin console. The same behaviour appears for all inbound calls.
Any ideas where to start looking? Also is there any way to get realtime logs of what's going on or to test a call plan in the same way that you'd use the asterisk CLI? for example:
asterisk -r
show dialplan
etc etc?
Thanks,
Chris
Just installed the 3CX Linux iso as a VM. I've got outbound & internal calls working OK, but having problems with inbound routes. When an inbound call occurs the phone rings for a fraction of a second then hangs up.
This occurs for the main trunk line, and all other DIDs that have been setup for ring groups, individual extensions etc.
If I test ringing a single extension that has no call forwards in place...
In the 3CX admin page I see:
07/13/2017 9:26:15 AM - [CM503003]: Call(C:5): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 192.168.30.110:2051
If I tcpdump on the phone and download the PCAP, I can see that the PBX itself is sending the cancel:
Session Initiation Protocol (CANCEL)
Request-Line: CANCEL sip:[email protected]:2057;line=xoq4bnoa SIP/2.0
Method: CANCEL
Request-URI: sip:[email protected]:2057;line=xoq4bnoa
Request-URI User Part: 402
Request-URI Host Part: 192.168.30.60
Request-URI Host Port: 2057
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.30.11:5060;branch=z9hG4bK-524287-1---f2039d7c41957367;rport
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]:5060;nf=e>;tag=34ad3a54
Call-ID: yNutAUDrMPjlFsLvgufmJw..
CSeq: 1 CANCEL
Sequence Number: 1
Method: CANCEL
User-Agent: 3CXPhoneSystem 15.5.1694.0 (1694)
Reason: SIP;cause=480;text="NO_ANSWER"
Reason protocols: SIP
Content-Length: 0
So this is a test from a mobile phone to a DID that goes directly to the extension in question, which shows up as being "green" in the admin console. The same behaviour appears for all inbound calls.
Any ideas where to start looking? Also is there any way to get realtime logs of what's going on or to test a call plan in the same way that you'd use the asterisk CLI? for example:
asterisk -r
show dialplan
etc etc?
Thanks,
Chris