Re: How does the 3CXPhone determine "Out of Office" for tunn
Okay, let me try approaching this problem from a different way. We have a test 3CX server, and we set up a test laptop. Both of them are doing basically nothing else at the moment except servicing this call: no other clients or programs running. The test laptop has Wireshark installed.
First, I made a call with the phone "Out of Office" (using the tunnel), and the call successfully went to voicemail (this was from 2627 to 5010). Then, I put the phone "In Office", told it to re-register, and made the same call. It said "Dialing" until it timed out.
I'm pasting the most recent entries in the Server Activity Log. Call #1 is the 16:21 call. Call #2 doesn't even show up at all, although it happened a minute or two afterward.
14-Jan-2015 16:21:29.091 Leg L:4.2[VMail] is terminated: Cause: BYE from PBX
14-Jan-2015 16:21:29.091 [CM503008]: Call(C:4): Call is terminated
14-Jan-2015 16:21:29.085 Leg L:4.1[Extn] is terminated: Cause: BYE from 172.18.3.179:5080
14-Jan-2015 16:21:22.129 [CM503007]: Call(C:4): VMail:9999 has joined, contact <sip:
[email protected]:40600>
14-Jan-2015 16:21:22.125 [CM503007]: Call(C:4): Extn:2627 has joined, contact <sip:
[email protected]:55781>
14-Jan-2015 16:21:22.123 L:4.2[VMail] has joined to L:4.1[Extn]
14-Jan-2015 16:21:21.974 [CM503025]: Call(C:4): Calling T:VMail:9999@[Dev:sip:
[email protected]:40600;rinstance=b86d629859d870be] for L:4.1[Extn]
14-Jan-2015 16:21:21.924 [CM503005]: Call(C:4): Forwarding: T:VMail:9999@[Dev:sip:
[email protected]:40600;rinstance=b86d629859d870be]
14-Jan-2015 16:21:21.924 L:4.1[Extn] forwards call from Extn:5010 to VMail:9999 based on rule Fwd[Available/NotReg]
14-Jan-2015 16:21:21.924 L:4.1[Extn] failed to reach Extn:5010, reason Not Registered
14-Jan-2015 16:21:21.924 [CM503017]: Call(C:4): Target is not registered: Extn:5010
14-Jan-2015 16:21:21.924 [CM503001]: Call(C:4): Incoming call from Extn:2627 to <sip:
[email protected]:5060>
14-Jan-2015 16:21:06.798 Leg L:3.2[VMail] is terminated: Cause: BYE from PBX
14-Jan-2015 16:21:06.798 [CM503008]: Call(C:3): Call is terminated
14-Jan-2015 16:21:06.787 Leg L:3.1[Extn] is terminated: Cause: BYE from 172.18.3.179:5080
14-Jan-2015 16:21:02.675 [CM503007]: Call(C:3): VMail:9999 has joined, contact <sip:
[email protected]:40600>
14-Jan-2015 16:21:02.672 [CM503007]: Call(C:3): Extn:2627 has joined, contact <sip:
[email protected]:55781>
14-Jan-2015 16:21:02.670 L:3.2[VMail] has joined to L:3.1[Extn]
14-Jan-2015 16:21:02.439 [CM503025]: Call(C:3): Calling T:VMail:9999@[Dev:sip:
[email protected]:40600;rinstance=b86d629859d870be] for L:3.1[Extn]
14-Jan-2015 16:21:02.390 [CM503005]: Call(C:3): Forwarding: T:VMail:9999@[Dev:sip:
[email protected]:40600;rinstance=b86d629859d870be]
14-Jan-2015 16:21:02.390 L:3.1[Extn] forwards call from Extn:5010 to VMail:9999 based on rule Fwd[Available/NotReg]
14-Jan-2015 16:21:02.390 L:3.1[Extn] failed to reach Extn:5010, reason Not Registered
14-Jan-2015 16:21:02.390 [CM503017]: Call(C:3): Target is not registered: Extn:5010
14-Jan-2015 16:21:02.388 [CM503001]: Call(C:3): Incoming call from Extn:2627 to <sip:
[email protected]:5060>
14-Jan-2015 16:20:55.087 [CM504001]: Endpoint Extn:2627: new contact is registered. Contact(s): [sip:
[email protected]:58535 / 2627,sip:
[email protected]:55781 / 2627]
The wireshark captures for both calls is attached. What I'm most suspicious of is the fact that for the unsuccessful call the SIP INVITE appears like this (and you can see that it keeps trying for about 30 seconds).
17 1.816859000 192.168.97.36 172.18.3.179 SIP/SDP 1092 Request: INVITE sip:
[email protected]:5060 |
and the successful call SIP INVITE looks like this:
24 4.808329000 192.168.97.36 172.18.3.179 SIP/SDP 1120 Unknown request: &\004\331\373Sip0INVITE sip:
[email protected]:5060 |
I can't tell what this signifies. Is it just because the Tunnel adds a header that Wireshark doesn't know how to interpret?
Can anyone see clues here as to what is going wrong? We can't see how anything on the network could be blocking anything, but I also can't think of any other explanation to why the SIP INVITES aren't making it to the server. Port 5060, which I think is the port being used, is open between the server and the laptop: I just verified that with telnet. Are there other ports I should verify?