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When someone hangs up after being on message bank the line is held and therefore during the night it can hold out every line.. and also the messages left are huge once we physically make the disconnection.


I have set up Reverse Polarity but this doesn't seem to change the situation. Has anyone got an idea of how to stop this....

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Well, that is one reason why every dialog should eventually hang up - at the very maximum after two hours. If the call is still occupying the line after this there is some more fundamental trouble.


Hangup detection is really a big problem in FXO. Every country does it differently. What country are you in?

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  • 1 month later...



I'm in South Africa and are experiencing the same thing except that the CS410 does not hang up at all, wether on a call and the other party has disconnected the call or if i have disconnected the call.


I'm also using the Windows based PBXNSIP together with a PATTON SMARTNODE 4828, and on this system if I do a disconnect it works perfectly. Below are the tone codes that we used for the PATTON:


profile call-progress-tone ZA_Dialtone

play 1 0 400 -13


profile call-progress-tone ZA_Dialtone

play 1 300 425 0 450 0


profile call-progress-tone ZA_Alertingtone

play 1 400 400 -13

pause 2 200

play 3 400 400 -13

pause 4 2000


profile call-progress-tone ZA_Busytone

play 1 500 400 -13

pause 2 500


profile call-progress-tone ZA_Releasetone

play 1 250 400 -13

pause 2 250


profile call-progress-tone ZA_Holdtone

play 1 30 400 -13

pause 2 60

play 3 30 400 -13

pause 4 4880


Please can you let me know how I can configure these tones on the CS410 to work for our country.



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Can you tell us exactly what this means?

play 1 250 400 -13

pause 2 250


We are working with the chip manufacturer to be able to set the disconnect tone in different countries. Do you know if the lines there use current disconnect or reverse polarity? Do you know what type of switches are in use? i.e. ericson, siemens?


If you route the calls to the auto attendant there is a Hangup Timeout that you can set to say 60 seconds so if all calls are routed to the AA and they don't go to an extension it should hang it up. Not pretty but a workaround until we can figure our the reorder tone or the disconnect detection mechanism.

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Here is an easy way to get a reorder tone so we can analyze it and figure out what the frequency and the duration is so we can configure the sipfxo driver to recongize the tone.



1. Set "Maximum VoiceMail Duration" to 30 seconds:

Select localhost domain, under Settings, you will see "Maximum

VoiceMail Duration".

2. Dial in from PSTN. When Auto Attendant answers, the extension that

did not currently online, then you can leave message. For example, 45.

3. When you hear VoiceMail IVR and start to record, hang be up your

PSTN phone (now the busy tone will be recorded).

4. Wait until the 30 seconds timeout, the line will be dropped.

5. Login as the extension 45 to check mail. (Select Settings -> Lists

-> Mailbox) Listen to the voice mail and see if the busy tone is

recorded. Right click on the voicemail and save the file to PC.

6. Send the recorded file to us.

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