Solved Unintended Routing to Conference Line

[MaxL]

I have a very unusual issue in my 3CX hosted PBX. When one of my agents calls 712-794-5270 or any number with a 712 area code, it routes them to the built in conference line. The problem is, this is happening intermittently. They could call this number, reach the conference line, hang up, call again, and reach the intended person. This also only happens on the Windows desktop client, it never happens through the web browser. Below is a sample activity log entry when I tested this call myself and reached the conference line. (FYI, our conference line extension is 900) Any ideas?

P.S. I have censored private info in this log entry which is why there are so many Xs


12/02/2021 6:23:31 PM - Leg L:9249.2[Line:10000>>17127945270] is terminated: Cause: BYE from local
12/02/2021 6:23:31 PM - [CM503008]: Call(C:9249): Call is terminated
12/02/2021 6:23:31 PM - Leg L:9249.1[Extn:101] is terminated: Cause: BYE from 127.0.0.1:5080
12/02/2021 6:23:26 PM - [CM503007]: Call(C:9249): Extn:101 has joined, contact <sip:[email protected]:5060/UDP>
12/02/2021 6:23:26 PM - [CM503007]: Call(C:9249): Line:10000>>17127945270 has joined, contact <sip:[email protected]:0/UDP>
12/02/2021 6:23:26 PM - L:9249.2[Line:10000>>17127945270] has joined to L:9249.1[Extn:101]
12/02/2021 6:23:26 PM - [CM505003]: Provider:[SIP.US] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [] PBX contact: [sip:XXXXXXX@XXXXXX:5060]
12/02/2021 6:23:26 PM - [CM503002]: Call(C:9249): Alerting Line:10000>>17127945270 by contact <sip:[email protected]:0/UDP>
12/02/2021 6:23:23 PM - [CM503025]: Call(C:9249): Calling T:Line:10000>>17127945270@[Dev:sip:[email protected]] for L:9249.1[Extn:101]
12/02/2021 6:23:23 PM - [CM503027]: Call(C:9249): From: Extn:101 ("XXX XXXXXX" <sip:[email protected]:0>) to T:Line:10000>>17127945270@[Dev:sip:[email protected]]
12/02/2021 6:23:23 PM - [CM503004]: Call(C:9249): Route 1: from L:9249.1[Extn:101] to T:Line:10000>>17127945270@[Dev:sip:[email protected]]
12/02/2021 6:23:23 PM - Line limit check: Current # of calls for line Lc:10000(@SIP.US[<sip:[email protected]:0/UDP>]) is 2; limit is 6
12/02/2021 6:23:23 PM - Call(C:9249): Call from Extn:101 to 7127945270 matches outbound rule 'Test'
12/02/2021 6:23:23 PM - [Flow] Call(C:9249): has built target endpoint: Out#:>>Rule{Test}>>7127945270 for call from L:9249.1[Extn:101]
12/02/2021 6:23:23 PM - [Flow] Target endpoint for 7127945270 is Out#:>>Rule{Test}>>7127945270
12/02/2021 6:23:23 PM - [CM503010]: Call(C:9249): Making route(s) from Extn:101 to <sip:[email protected]:0/UDP>
12/02/2021 6:23:23 PM - [CM505001]: Endpoint Extn:101: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXPhone for Windows 16.3.0.264] PBX contact: [sip:[email protected]:5060]
12/02/2021 6:23:23 PM - [CM500002]: Call(C:9249): Info on incoming INVITE from Extn:101:
Invite-IN Recv Req INVITE from 127.0.0.1:5080 tid=Pjb847292e679e4a1dbb0744655d4bb196 Call-ID=b8ca4fa88e9a4efba40fa163ae536679:
INVITE sip:[email protected]:5060;transport=UDP;tag3cx=7d1983e5337340a3b02e2cfbbbbe68ba SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bK-524287-2---Pjb847292e679e4a1dbb0744655d4bb196;rport=5080
Via: SIP/2.0/UDP [::ffff:xxxxxxxx]:56392;branch=z9hG4bK-524287-1---tunneltid;rport;tnlid=clnt.0-4b23d2fe7f6349809a9e2e97fbe39b71
Via: SIP/2.0/TCP 192.168.1.196:50195;rport;branch=z9hG4bKPjb847292e679e4a1dbb0744655d4bb196;alias
Max-Forwards: 69
Record-Route: <sip:[email protected]:5080;user=proxy;uri=clnt.0-4b23d2fe7f6349809a9e2e97fbe39b71>
Contact: "XXX XXXXXX" <sip:[email protected]:5060;rinstance=0-4b23d2fe7f6349809a9e2e97fbe39b71>
To: <sip:[email protected];tag3cx=7d1983e5337340a3b02e2cfbbbbe68ba>
From: "XXX XXXXXXXX"<sip:[email protected]>;tag=fe029530a4d84c5f9a94e03b55c0ace7
Call-ID: b8ca4fa88e9a4efba40fa163ae536679
CSeq: 28521 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="xga9CoCJRv",realm="3CXPhoneSystem",nonce="414d535961a9556b72:51e51e2380637b0b7eb006cc7b369f33",uri="sip:[email protected];transport=tcp;tag3cx=7d1983e5337340a3b02e2cfbbbbe68ba",response="ecd3085fcc9048a131844e4d32004927",algorithm=MD5
Supported: replaces, 100rel, timer, norefersub
User-Agent: 3CXPhone for Windows 16.3.0.264
Content-Length: 461

v=0
o=- 3847458204 3847458204 IN IP4 192.168.1.196
s=pjmedia
b=AS:117
t=0 0
a=X-nat:0
m=audio 10010 RTP/AVP 0 8 9 18 120 96
c=IN IP4 192.168.1.196
b=TIAS:96000
b=AS:117
a=rtcp:10011
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
m=video 0 RTP/AVP 31
c=IN IP4 127.0.0.1
12/02/2021 6:23:23 PM - [CM503001]: Call(C:9249): Incoming call from Extn:101 to <sip:[email protected]:0>

 

[NickD_3CX]

Hi!

When you say they reach a Conference Line, I presume you mean the 3CX Conference Server right?
By default, the conference server number on a 3-digit 3CX installation is 700.

Can you confirm that you have not changed it to let's say 712?
You can check this from Settings --> Conferencing, --> Audio tab.

I am thinking that maybe the device they are calling from has a "time out", so the agent might dial "712", then pause for a few seconds reading the numberm and before they have time to continue typing, the device starts auto-dialing the number that is already typed in.

 

[MaxL]

that was my first thought, so I changed the conference server number to 900. The problem still occurred after that change.

 

[MaxL]

What would be even better is if we could disable the conference line service all together. We are not using this service as we already have a 3rd party conference service in place.

 

[ChrisC_3CX]

To gather a bit more information on this and hopefully figure out what's happening:

that was my first thought, so I changed the conference server number to 900. The problem still occurred after that change.

Just to confirm, was the conference extension indeed set to 712 before?

We are not using this service as we already have a 3rd party conference service in place.

What's the configuration involving this 3rd part conference service? Is it running locally on the 3CX Host or do you have to call an external number via the SIP Trunk?

12/02/2021 6:23:23 PM - [Flow] Call(C:9249): has built target endpoint: Out#:>>Rule{Test}>>7127945270 for call from L:9249.1[Extn:101]

For the call you provided logs for, I see it's matching an Outbound Rule named "Test". Is this your normal Outbound rule used for every call? If not, what's different about this outbound rule, is there any chance the issue has something to do with the Outbound Rule being matched?


When you face such an issue, do you actually see the call connected to the conference extension if you check under "Dashboard >> Number of Calls in use":

 

[MaxL]

I will answer your questions in order.

The extension was set to 700 before which I believe is the default.

This 3rd party conference service is completely unrelated and has no connection to 3CX.

This test outbound rule is a carbon copy of the production one just with a different DID as the outbound caller ID.

 

[ChrisC_3CX]

This 3rd party conference service is completely unrelated and has no connection to 3CX.

It may not be related to 3CX but eliminating the possibility of it being related to this scenario is vital. Knowing the basics of how it works and is utilized should be enough.

When you face such an issue, do you actually see the call connected to the conference extension if you check under "Dashboard >> Number of Calls in use":

Have you happened to check the "Number of Calls in use" section on the Dashboard while this is happening? What do you see?

 

[MaxL]

Thanks Chris. To your first response. When I say it is un-related, I mean it physically has no connection to 3CX whatsoever. We utilize a completely different platform and application for our conferencing solutions. It is simply not possible that this is affected in any way.

I have noticed while looking at the "calls in use" that the number never went above 3 active calls. We have a total of 8 SIM calls that can go through our PBX and SIP trunk at any given time.

 

[ChrisC_3CX]

I have noticed while looking at the "calls in use" that the number never went above 3 active calls. We have a total of 8 SIM calls that can go through our PBX and SIP trunk at any given time.

Have you also checked if you can actually see the call connected to the Conferencing extension as shown here:
https://www.3cx.com/community/threads/unintended-routing-to-conference-line.86227/post-406456

 

[MaxL]

I had not though to check my active calls list to see if it shows up as such. It looks like it might be my SIP Trunk provider doing this strange routing by this listing. I might reach out to them to see if they can find out what is going on.

 

[ChrisC_3CX]

I might reach out to them to see if they can find out what is going on.

I think that would be your best bet as the call does not seem to be connected to your PBX's conference extension but rather, to an external number via your SIP Provider. Let us know how it goes!

 

[MaxL]

Looks like this ended up being my SIP trunk provider. They have made routing adjustments and the problem has resolved. Thanks for the help.

 

[ChrisC_3CX]

Well I'm happy to hear you managed to identify the culprit!

Feel free to start a new thread should anything else pop up!