Calls terminated at approximately 3 hours

[jdcali]

I noticed open calls are terminated every 3 hours. Is this intentional, a bug or something that can be resolved in the config file?

 

[nb]

Hi

No it shouldn't be the case. Do you really make 3 Hours connected on this call? What happens then? It disconnects?

Are you recording the calls my any chance?

 

[jdcali]

No I'm not recording I'm just streaming audio from one computer on my LAN to another and it disconnects every 3 hours. I tried doing this skype, teamspeak, ventrilo and I never had problems with disconnects.

The next time it happens I'll post the server log.

Hi

No it shouldn't be the case. Do you really make 3 Hours connected on this call? What happens then? It disconnects?

Are you recording the calls my any chance?

 

[nb]

Make sure they are in verbose logs and make sure that you also post media server trace logs.

By streaming audio what do you mean exactly? Making a normal call to another entity?

Because you are right - all the calls shown in the history all disconnected at 3HOUR timestamp.

Is it actually making 3hrs connected? You can confirm that?

Also can you eliminate the Voip phone and try making the same test with Xlite for example? So we can identify whether it is related to the Voip phone or from the Pbx? Thanks

 

[nb]

Hi

I double checked and in fact yes - calls will be terminated after 3 hours - there is a value in the PBX that controls the maximum call duration for each call.

However you can do the following:

Go to Settings, advanced , custom parameters
(BE CAREFULL HERE - this page is sensitive)
go to the bottom and add the following:

MAXCALLDURATION

Description: leave blank

Value - default is 10800 sec. modify this to how long you want it - Note in seconds.

This globally controls maximal call duration. All calls (even if call is alive) will be terminated after specified timeout.

 

[jdcali]

Sorry I don't know how to access the media server trace logs but I've included the Server Activity Log. By the way, the recent disconnect happened at 15:50:51.062 and I reconnected at 15:54:04.328

By streaming audio what do you mean exactly? Making a normal call to another entity?

Is it actually making 3hrs connected? You can confirm that?

Yes I'm making a normal call from 1 computer to another on a lan and the server is also hosted locally. What I mean by streaming audio is that I have my mic next to a radio and after 3 hours of the line being open the (server?) disconnects the call on both ends.

16:19:15.656 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

15:59:15.546 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

15:54:04.921 [CM503007]: Call(38): Device joined: sip:[email protected]:57250;rinstance=c8f669f2414a7fa7

15:54:04.906 [CM503007]: Call(38): Device joined: sip:[email protected]:1519;rinstance=50be11f760b70a9e

15:54:04.906 [CM505001]: Ext.11: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.0.6061.0] Transport: [sip:192.168.1.2:5060]

15:54:04.906 [CM503002]: Call(38): Alerting sip:[email protected]:57250;rinstance=c8f669f2414a7fa7

15:54:04.328 [CM503004]: Call(38): Calling: Ext:Ext.11@[Dev:sip:[email protected]:57250;rinstance=c8f669f2414a7fa7]

15:54:04.328 [CM503010]: Making route(s) to <sip:[email protected]:5060>

15:54:04.328 [CM505001]: Ext.10: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.0.6061.0] Transport: [sip:192.168.1.2:5060]

15:54:04.328 [CM503001]: Call(38): Incoming call from Ext.10 to <sip:[email protected]:5060>

15:50:51.062 [CM503008]: Call(37): Call is terminated

15:50:51.062 [CM503008]: Call(37): Call is terminated

15:50:51.062 [CM503008]: Call(37): Call is terminated

15:39:15.421 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

15:19:15.312 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

14:59:15.203 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

14:39:15.062 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

14:19:14.890 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

13:59:14.796 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

13:39:14.671 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

13:29:52.296 [CM504010]: Fax Service: unregistered contact sip:88:5100;user=phone

13:29:51.781 [CM504010]: Fax Service: unregistered contact sip:88:5100;user=phone

13:29:34.656 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

13:24:34.406 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

13:19:34.218 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

13:19:13.531 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

13:14:34.031 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

13:09:33.843 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

13:04:33.609 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:59:33.406 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:59:13.359 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

12:54:33.218 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:50:52.171 [CM503007]: Call(37): Device joined: sip:[email protected]:57250;rinstance=c8f669f2414a7fa7

12:50:52.171 [CM503007]: Call(37): Device joined: sip:[email protected]:1519;rinstance=50be11f760b70a9e

12:50:51.656 [CM505001]: Ext.11: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.0.6061.0] Transport: [sip:192.168.1.2:5060]

12:50:51.656 [CM503002]: Call(37): Alerting sip:[email protected]:57250;rinstance=c8f669f2414a7fa7

12:50:51.078 [CM503004]: Call(37): Calling: Ext:Ext.11@[Dev:sip:[email protected]:57250;rinstance=c8f669f2414a7fa7]

12:50:51.078 [CM503010]: Making route(s) to <sip:[email protected]:5060>

12:50:51.078 [CM505001]: Ext.10: Device info: Device Identified: [Man: 3CX Ltd.;Mod: 3CXVoipPhone;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [3CXVoipPhone 3.0.6061.0] Transport: [sip:192.168.1.2:5060]

12:50:51.062 [CM503001]: Call(37): Incoming call from Ext.10 to <sip:[email protected]:5060>

12:49:57.578 [CM503008]: Call(36): Call is terminated

12:49:57.578 [CM503008]: Call(36): Call is terminated

12:49:57.578 [CM503008]: Call(36): Call is terminated

12:49:33.000 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:44:32.812 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:39:32.609 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:39:13.203 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

12:34:32.421 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:29:32.218 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:24:32.031 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:19:31.828 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:19:13.093 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

12:14:31.625 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:09:31.453 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

12:04:31.218 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:59:31.046 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:59:12.984 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

11:54:30.859 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:49:30.609 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:44:30.453 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:39:30.250 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:39:12.812 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

11:34:30.109 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:29:29.890 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:24:29.703 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:19:29.500 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:19:12.718 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

11:14:29.296 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:09:29.109 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

11:04:28.921 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

10:59:28.718 [CM504008]: Fax Service: registered as sip:[email protected]:5060 with contact sip:[email protected]:5100;user=phone

10:59:12.593 [CM506001]: STUN request to resolve SIP external IPort mapping is sent to STUN server 75.xxx.xxx.xxx:3478 over Transport 192.168.1.2:5060

Make sure they are in verbose logs and make sure that you also post media server trace logs.

By streaming audio what do you mean exactly? Making a normal call to another entity?

Because you are right - all the calls shown in the history all disconnected at 3HOUR timestamp.

Is it actually making 3hrs connected? You can confirm that?

Also can you eliminate the Voip phone and try making the same test with Xlite for example? So we can identify whether it is related to the Voip phone or from the Pbx? Thanks

 

[jdcali]

I will give this a try.

Thank You!

Hi

I double checked and in fact yes - calls will be terminated after 3 hours - there is a value in the PBX that controls the maximum call duration for each call.

However you can do the following:

Go to Settings, advanced , custom parameters
(BE CAREFULL HERE - this page is sensitive)
go to the bottom and add the following:

MAXCALLDURATION

Description: leave blank

Value - default is 10800 sec. modify this to how long you want it - Note in seconds.

This globally controls maximal call duration. All calls (even if call is alive) will be terminated after specified timeout.

 

[Vali_3CX]

No I'm not recording I'm just streaming audio from one computer on my LAN to another and it disconnects every 3 hours.

Hi

To change a little the subject - I'm interested if there is (not) a bug into the Voip Phone: you said you didn't recorded calls, however, as I can see in the screenshot you attached, almost all calls appears to be recorded (see below, "Reco[rded]..." column, "Yes")

So, into the the Voip Phone's File/Recordings... dialog, please check if "Record as local files" option is checked. If is, then all recorded calls should be available on that list (of course, if you don't deleted them meantime). You can check also directly the recordings storage folder on user's profile folder, usually
C:\Documents and Settings\jdcali\Local Settings\Application Data\3CX VoIP Phone\Recordings

If "Record as local files" option is not checked, then all recorded calls should be available on the PBX machine, on the specific folder.

Regards
vali

 

[LeonidasG_3CX]

CDR Commands.

MAXCALLDURATION(Min value is MAXNOANSWERTIMEOUT+180 sec, default 10800 sec = 3 hours)



if you make calls longer than 3 hours you can change this value to whatever you like.