when i hung up, phone is going on ringing :-(

[Tugi2o]

but the rfc guideline say 3xc have only to send a "CANCEL" if there was no response if there was any response there is no need for "CANCEL" If any Response came frome the other side like it indicate that it is ringing the option for end this SESSION is "BYE".
it would be a workaround for the mistakes of other people when 3cx would send both "CANCEL" & "BYE" to end the session.

3cx may solve this problem by extending their software with a further option, where the user can set the signal which is sent when i hang up the phone.

why does 3cx PHONE send the right signal and 3cx PHONE SYSTEM doesn't?

i think that 3cx is the best phone system software, but we cannot use it because of this little "problem" which could be easily solved -> i hope this happens soon.

 

[laurian3000]

Same problem here with my VOIP Provider

Please Help, I want to switch to 3CX but I'm concerning about a problem:

I'm using 3CX with my Eurovoice account and everything works fine, but if the party doesn't respond and I hang up the phone, 3CX doesn't send correctly sends the SIP command CANCEL to hang Up the operator line or the Voip Provider doesn't support BYE


If I register my VOIP account in my SIP Phone Linksys SPA941or any SoftPhone everything works fine , but any of the soft and hardware phones I tried sends the command CANCEL ( not BYE) and the provider understand it and close the line.

I checked the SIP traffic with Etheral, (dialing my Mobile # and then hang up) and I found the following SIP traffic:
-----------------------------------------
1st scenario) from 3CX PBX to my mobile nr through Eurovoice.ro when the Provider doesn hangUp the line(where EuroVoice Proxy=193.231.78.10, 3CX Pbx=192.168.1.10 and my phone SPA941=192.168.1.12)
No. Time Source Destination Protocol Info
1 14:35:29.831847 192.168.0.14 192.168.1.10 SIP Request: REGISTER sip:192.168.1.10:6060
2 14:35:29.934907 192.168.1.10 192.168.0.14 SIP Status: 200 OK (1 bindings)
4 14:35:30.794898 192.168.1.12 192.168.1.10 SIP Request: REGISTER sip:192.168.1.10:6060
5 14:35:30.897779 192.168.1.10 192.168.1.12 SIP Status: 200 OK (1 bindings)
6 14:35:31.886832 192.168.1.12 192.168.1.10 SIP/SDP Request: INVITE sip:[email protected]:6060, with session description
7 14:35:31.990767 192.168.1.10 192.168.1.12 SIP Status: 407 Proxy Authentication Required
8 14:35:32.006209 192.168.1.12 192.168.1.10 SIP Request: ACK sip:[email protected]:6060
9 14:35:32.020929 192.168.1.12 192.168.1.10 SIP/SDP Request: INVITE sip:[email protected]:6060, with session description
12 14:35:32.124258 192.168.1.10 192.168.1.12 SIP Status: 100 Trying
17 14:35:32.227948 192.168.1.10 193.231.78.10 SIP/SDP Request: INVITE sip:[email protected]:5060, with session description
18 14:35:32.244219 193.231.78.10 192.168.1.10 SIP Status: 407 Proxy Authentication Required
19 14:35:32.246510 192.168.1.10 193.231.78.10 SIP Request: ACK sip:[email protected]:5060
20 14:35:32.349082 192.168.1.10 193.231.78.10 SIP/SDP Request: INVITE sip:[email protected]:5060, with session description
21 14:35:32.372722 193.231.78.10 192.168.1.10 SIP Status: 100 Trying
26 14:35:32.452491 193.231.78.10 192.168.1.10 SIP Status: 100 trying -- your call is important to us
28 14:35:33.964248 192.168.1.10 193.231.78.10 SIP Request: REGISTER sip:eurovoice.ro
29 14:35:33.977356 193.231.78.10 192.168.1.10 SIP Status: 200 OK (6 bindings)
35 14:35:37.133116 193.231.78.10 192.168.1.10 SIP Status: 180 Ringing
36 14:35:37.235573 192.168.1.10 192.168.1.12 SIP Status: 180 Ringing
38 14:35:41.109845 192.168.1.12 192.168.1.10 SIP Request: CANCEL sip:[email protected]:6060
39 14:35:41.217222 192.168.1.10 192.168.1.12 SIP Status: 200 OK
40 14:35:41.217294 192.168.1.10 192.168.1.12 SIP Status: 487 Request Terminated
41 14:35:41.217345 192.168.1.10 193.231.78.10 SIP Request: BYE sip:[email protected]
42 14:35:41.241555 192.168.1.12 192.168.1.10 SIP Request: ACK sip:[email protected]:6060
43 14:35:41.745594 192.168.1.10 193.231.78.10 SIP Request: BYE sip:[email protected]
47 14:35:42.751478 192.168.1.10 193.231.78.10 SIP Request: BYE sip:[email protected]
48 14:35:44.763201 192.168.1.10 193.231.78.10 SIP Request: BYE sip:[email protected]

---------------------------------------------
2 scenario)from my SPA941 directly through EuroVoice when the line is correctly hang-up (where 193.231.78.10 is the SIP Proxy of the provider and 192.168.1.10 my phone)

No. Time Source Destination Protocol Info
11 14:28:53.919827 192.168.1.10 193.231.78.10 SIP/SDP Request: INVITE sip:[email protected], with session description
12 14:28:53.934027 193.231.78.10 192.168.1.10 SIP Status: 407 Proxy Authentication Required
13 14:28:53.934665 192.168.1.10 193.231.78.10 SIP Request: ACK sip:[email protected]
14 14:28:53.935349 192.168.1.10 193.231.78.10 SIP/SDP Request: INVITE sip:[email protected], with session description
15 14:28:53.959244 193.231.78.10 192.168.1.10 SIP Status: 100 Trying
16 14:28:54.037698 193.231.78.10 192.168.1.10 SIP Status: 100 trying -- your call is important to us
34 14:28:58.685348 193.231.78.10 192.168.1.10 SIP Status: 180 Ringing
36 14:29:00.152449 192.168.1.10 193.231.78.10 SIP Request: CANCEL sip:[email protected]
37 14:29:00.170042 193.231.78.10 192.168.1.10 SIP Status: 200 canceling
38 14:29:00.204304 193.231.78.10 192.168.1.10 SIP Status: 487 Request Terminated
39 14:29:00.204949 192.168.1.10 193.231.78.10 SIP Request: ACK sip:[email protected]
------------------------------------------------

As you can see in the 1st scenariu (through 3CX BX) when I hang up the phone the phone sends the CANCEL message to 3CX, and 3CX sends BYE to another IP 193.231.78.24 that is possible to be their media server , NOt to the Ip of the proxy 193.231.78.10. As the phone sends the CANCEL


Any help will be appreciated.

 

[MAuVE]

I tried 3CX for a particular setup but hit onto the same problem i.e the called phone keeps ringing after the caller hungs-up.

My setup:

One Cisco 1751-V running CallManager Express with some SCCP phones (7905, 7912, 7920, 7970)

Via sip-ua I connect to my sip provider for outgoing calls.

Incoming calls are ISDN and POTS.

Everything works ok for more that 1,5 years.



I tried 3CX because I wanted to add a second sip trunk line and CME only accepts one sip-ua.

Sip provider asigns only one line per account.

So 3CX plays the role of a back to back user agent.

It gets registered to two trunk accounts on the outgoing side and accepts all the internal phones of CME as extensions

Everything works, but the known problem.

The same happens with a cisco ATA 188 directly registered on 3CX, so do not waste your time questioning the interoperability of CME with 3CX.

The problem is between 3CX and sip provider

 

[archie]

I think we have solved this problem in current version. Now we're using CANCEL instead of BYE to terminate unestablished calls. We're testing this build now, and it will be available pretty soon.

 

[ITWorks]

I now have the "calls go on ringing after called from 3CX through Grandstream GXW400x Gateway" problem. Did not have the problem until upgrading to 3CX Phone System Version 3.0.2295.0 and updating the Grandstream Firmware 1.0.0.45.

In another post jaiume wrote:

Grandstream have told me they will be changing their firmware to terminate a call on BYE without CANCEL, but they said that its not really correct w.r.t. the protocol.

Archie wrote:

I think we have solved this problem in current version. Now we're using CANCEL instead of BYE to terminate unestablished calls. We're testing this build now, and it will be available pretty soon.

Could this be the source of my new problem?

Mark

 

[Anonymous]

Actually it might be just a one off.

But I had the same with an AT530 phone, I will test a bit more but the phone did not disconnect either. I had to power down the phone to get the phone to realise that the call was terminated.

Weird

 

[ITWorks]

Calls fail to hang up

This is still a problem.

The phone (10.10.30.70) sends a BYE to 3CX server (10.10.30.3).
The 3CX server sends a BYE to the Grandstream GXW4004 FXS Gateway (10.10.30.10). However the Gateway does not disconnect the call and the phone goes on ringing. This was not a problem until the change to the latest versions of software as indicated my previous post.

"10.10.30.70", "10.10.30.3", "SIP", "Request: BYE sip:[email protected]:5060"
"10.10.30.3", "10.10.30.70", "SIP", "Status: 200 OK"
"10.10.30.3", "10.10.30.10", "SIP", "Request: BYE sip:[email protected]:5060"
"10.10.30.10", "10.10.30.3", "SIP", "Status: 200 OK"

Was there really a change from BYE to CANCEL used to terminate calls, as Archie had indicated previously in this new version of 3CX?