Incoming Call hangs up, AudioCode MP118 FXO does not

[Andy Schmidt]

Hi,

If I dial into the PSTN, I get my automated attendant (Ext. 70). Soon after I hear the voice, I hang up.
Apparently, 3CX never knows that the call terminated and continues to wait for input for 15 seconds, then does the timeout handling (Ring Group 89). When I pick up that call, I have Verizon saying "if you would like to place a call..."

12:29:06.366 Call::Terminate [CM503008]: Call(54): Call is terminated
12:29:06.350 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70
12:29:06.350 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70
12:29:06.335 Call::Terminate [CM503008]: Call(54): Call is terminated
12:29:06.335 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70
12:29:00.632 CallCtrl:nLegConnected [CM503007]: Call(54): Device joined: sip:[email protected]:5060;transport=udp
12:28:59.100 CallCtrl:nSelectRouteReq [CM503004]: Call(54): Calling: RingAll:89@[Dev:sip:[email protected]:5060;transport=udp, Dev:sip:[email protected]:5060;transport=udp]

12:28:12.257 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70
12:28:12.241 CallCtrl:nLegConnected [CM503007]: Call(54): Device joined: sip:[email protected]:5060
12:28:11.960 CallCtrl:nLegConnected [CM503007]: Call(54): Device joined: sip:
12:28:11.960 CallCtrl:nSelectRouteReq [CM503004]: Call(54): Calling: IVR:70@[Dev]
12:28:11.944 Line:rintEndpointInfo [CM505002]: Gateway:[PSTN] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Audiocodes-Sip-Gateway-MP-118 FXO/v.5.00A.024] Transport: [sip:63.107.174.130:5060]
12:28:11.944 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70
12:28:11.944 CallCtrl:nIncomingCall [CM503001]: Call(54): Incoming call from 2019349213@(Ln.10800@PSTN) to [sip:[email protected]:5060]
12:28:11.929 LineCfg::getInboundTarget [CM503011]: Inbound office hours' rule for LN:10800 forwards to DN:70

 

[N_G]

Its probably because your audiocodes gateway has not been configured correctly to detect the hang up. Then it doesnt send a BYE request to 3CX and therefore the line has to time out before it is hang up. This is an audiocodes/gateway configuration issue.....

 

[archie]

What version of 3CX PBX are you using?

 

[Andy Schmidt]

Hi,

thanks for the response.

Version is 5.1.4393.0.

I've done another test, watching the line indicator light on the MP118.

- When I dial from the outside (using my cell phone), the Line 1 light goes green (my only line during testing).
- It stays on while I listen to the 3CX automated attendant.
- Now the caller (my cell phone) hangs up.
- The line line light STAYS green on the MP118 FX. In 3CX the "line status" shows that line as "orange"
- If the 3CX automated attendant eventually times out and puts the call though to the operator, who then "hangs up", then the connection is finally dropped.
- if the operator does NOT pick up, then the line status in 3CX stays orange for the line and the extension, until it times out and both lights return to green.

However, I had ONE case, where the line stayed orange for 40 minutes and for the extension too - although when I access that extension it was NOT ringing, did not have any indication that a call was in progress (I could pick up the handset and get a dial-tone, etc).

In the Tel Profile Settings I have Polarity Reversal and Current Disconnect both enabled, dito in "General Parameter" under "Disconnect and Answer Supervision".

I agree that it might be a "configuration" issue, but the point is, I'm purchased "supported" devices and followed the configuration steps you posted for the device. Yet, I ran into at least 3 parameters that apparently ARE needed but were NOT included in your sample...?

 

[archie]

I agree that it might be a "configuration" issue, but the point is, I'm purchased "supported" devices and followed the configuration steps you posted for the device. Yet, I ran into at least 3 parameters that apparently ARE needed but were NOT included in your sample...?

We're supporting SIP end of your device, and as you can see SIP side works perfectly. We can not support Telco side of thing, especisally those which are regional dependant. It is not our fault that every country has its own standard of tone/pulse dialing, disconnect indication, service tones, etc.

 

[N_G]

For country specific settings of the AUDIOCODES gateway, you have to contact Audiocodes support or the reseller you bought it from. We are not the Audiocodes support department.

 

[Nebula]

I had the same problem with the 3CX generated config for a Patton 4112 FXO gateway. It is a very simple fix, you just need to adjust find the line of config that refers to the release tone and adjust the values to suit your region. With the Patton it is as follows..

"3CX generated config"
profile call-progress-tone defaultReleasetone
flush-play-list
play 1 400 400 -24
pause 2 350
play 3 225 400 -24
pause 4 525

For UK change this entry to

profile call-progress-tone defaultReleasetone
flush-play-list
play 1 1500 400 0

 

[Andy Schmidt]

>> find the line of config that refers to the release tone and adjust the values to suit your region <<

Nebula - thank's for the hint. I'll try to learn more about the release tone used in the U.S. (I'm not in the U.K.!)

>> It is not our fault that every country has its own standard of tone/pulse dialing, disconnect indication, service tones, etc. <<

Hm - may I respectfully suggest that the U.S. is a major market with a significant revenue potential. You should have a vested interest to supply working configurations for major markets. Hopefully, I'm not your first U.S. customers - so if anyone got your system to work in the U.S., wouldn't it make sense to collect the "regional" settings and have them ready to go (or set up) in your product so that more people will move from the "trial" stage into a "buy" relationship with you?

>> We are not the Audiocodes support department <<

Thank you for that constructive comment. I am aware that you are NOT the AudioCodes support department. However, your web site http://www.3cx.com/voip-gateways/audiocodes-mp.html lead me to believe that choosing "supported" devices for which you post configuration instructions would leave me with a WORKING system in a MAJOR market. In fact your page ends with the promise:
"Congratulations. The AudioCodes MP114 is now configured to make and receive calls using the 3CX Phone System."
Based on your response, I suggest a strong disclaimer instead: "These parameters will only get you started. If you live in unheard-of places, like the U.S., then additional local settings are required that every customer will have to figure out by themselves!"

 

[N_G]

Andy,

First of all the settings are not necessarily US wide - its dependent on your telco provider. When you follow our configuration guide, you have a working system with 3CX on the SIP side of things and with the default tone set configuration of the Audiocodes. The interaction with between the Audiocodes gateway and the telco lines is something you have to take up with Audio codes. They should provide you with the correct settings to use with the particular telco provider you are using. I am not sure why you insist on expecting 3CX to do Audiocodes support. And at that, at no cost at all. Its similar to asking Microsoft for support on the server hardware on which you installed Windows.

Note that the scope of this forum is to give pointers where to look - its not the equivalent of commercial level support and it would be unreasonable to expect that.

Therefore, based on your previous posts, we can conclude that the problem is that the Audiocodes does not detect a hang up, and therefore the most likely place to check is the dial tone and hang up tone settings on the Audiocodes device.

 

[Philco]

Andy, out of curiosity, what did you adjust in 3cx to make it work as it did look like it was an MP118 configuration.

If it was any regional settings that was required for teleco then I guess it was the MP118 that wasnt configured correctly and I wouldnt expect any info to be provided by 3cx on that appart from the parameters for efective communication between 3cx server and the MP118.

Just curious.

A just read Nicks reply too

Phil

 

[cornisland9]

>> find the line of config that refers to the release tone and adjust the values to suit your region <<

Nebula - thank's for the hint. I'll try to learn more about the release tone used in the U.S. (I'm not in the U.K.!)

>> It is not our fault that every country has its own standard of tone/pulse dialing, disconnect indication, service tones, etc. <<

Hm - may I respectfully suggest that the U.S. is a major market with a significant revenue potential. You should have a vested interest to supply working configurations for major markets. Hopefully, I'm not your first U.S. customers - so if anyone got your system to work in the U.S., wouldn't it make sense to collect the "regional" settings and have them ready to go (or set up) in your product so that more people will move from the "trial" stage into a "buy" relationship with you?

>> We are not the Audiocodes support department <<

Thank you for that constructive comment. I am aware that you are NOT the AudioCodes support department. However, your web site http://www.3cx.com/voip-gateways/audiocodes-mp.html lead me to believe that choosing "supported" devices for which you post configuration instructions would leave me with a WORKING system in a MAJOR market. In fact your page ends with the promise:
"Congratulations. The AudioCodes MP114 is now configured to make and receive calls using the 3CX Phone System."
Based on your response, I suggest a strong disclaimer instead: "These parameters will only get you started. If you live in unheard-of places, like the U.S., then additional local settings are required that every customer will have to figure out by themselves!"

I have the solution for audiocodes mp-114 and mp-118 not disconnecting calls. It is just one small enty in ini file..

 

[Nebula]

Care to share the answer??

 

[Andy Schmidt]

Hi,

well, the credit goes to Tommy Vaughn. He emailed me privately and shared his discovery from last week with me. I was sceptical - but tried tonight, and so far I have been unable to reproduce my problem, when before it was reproducible most of the time.

The key was THIS parameter:
CurrentDisconnectDuration = 500

Now, the ODD thing is, how the MP118 FXO reacted when I applied it.
First I applied it by retrieving the INI file from the Web Interface and then added the value to the [Analog Params] section, and then sent the INI file back to the device through the web interface. I allowed it to reboot, retrieved the INI file again - and my parameter was NOT there.

Then I stuck it below the:
ENABLECURRENTDISCONNECT = 1
in the SIP Params section.

I sent THAT INI file again, allowed the device to reboot. Then I retrieve the INI file again and this time the parameter had MOVED to the [Analog Params] section. But, not only that, suddenly my entire INI file had drastically changed. Suddenly countless parameters were missing (I'm assuming they may have been defaults).

I'm posting both my "before" and "after" INI files with this message.

So - in summary: Both Tommy and I placed our file into:
[SIP Params]
ENABLECURRENTDISCONNECT = 1
CurrentDisconnectDuration = 500

But after reboot, both HIS INI file and mine show it here:

[Analog Params]
CurrentDisconnectDuration = 500

I should mention that my location is NJ, US. My local telco is Verizon. The "no disconnect" problem seemed to be different, if the call was placed from a land line, vs. calling from a land line! Also, my INI file addresses the "CallerID" problem by waiting for the second ring (before I was NOT getting CallerID for cell phones).