I am facing a serious problem at an Asterisk installation. First of all, let me describe the topology: I have installed an OpenVox card with 4 FXO ports which are connected to the relative FXS ports on a modem/router given by a local provider. The problem is that when someone calls and redirected to an internal extension after hearing the IVR (or use the direct dialin) and hangs up before the callee answers the phone keep ringing for about 30-45 seconds (the amount of time that is configured in the general PBX settings). In fact even though it has been dropped for quite some time I watched through Asterisk that it is even unreasonably driven to the voicemail of the internal number that had been called.
I tried the busydetect and busycount but nothing changed. More importantly, I have noticed that the problem occurs only when the call is directed to an internal number after the IVR. When I change the inbound route and the call goes directly to any internal number without the intervention of the IVR then the problem does not incur. Based on this I am drawn to the conclusion that there is obviously something wrong with asterisk and not the provider.
In addition to the above, another deduction from the tests I made is that if the caller drops the call while the IVR message is in progress then the call (and the dahdi channel) is hanged up with a short perfectly expected delay. So the problem appears only when the call is transferred from the IVR to an internal number. Can anyone help me?
I don’t think it has anything to do with your IVR set up in dialplan. This seems to be a problem of disconnect supervision on analog lines.
Do you have callprogress set to yes in chan_dahdi.conf?
A little bit old but following link might help you resolve the issue. shyju.wordpress.com/tag/asteris … ess-tones/
What’s making the difference is that IVR is answering the call, so you need disconnect supervision for a calling part clear, whereas, with the direct routing, the call is never answered, and the cessation of ringing current will clear the call.
[quote=“satish4asterisk”]I don’t think it has anything to do with your IVR set up in dialplan. This seems to be a problem of disconnect supervision on analog lines.
Do you have callprogress set to yes in chan_dahdi.conf?
A little bit old but following link might help you resolve the issue. shyju.wordpress.com/tag/asteris … ess-tones/
–Satish Barot[/quote]
I have used call progress but nothing changed. I have read this useful topic but unfortunately there was any solution.
Analogue lines always cause problems with call supervision, but it sound like you don’t have a real analogue line. I suspect your inteface is only intended to be used by humans.
It is very unlikely that the device is providing any DC signalling of disconnect supervision. You will need to listen to it with an analogue phone to see if it provides any inband signalling.
If your application needs reliable supervision, you need to use ISDN or use SIP, via an ITSP, who will, in turn, use ISDN.
[quote=“david55”]Analogue lines always cause problems with call supervision, but it sound like you don’t have a real analogue line. I suspect your inteface is only intended to be used by humans.
It is very unlikely that the device is providing any DC signalling of disconnect supervision. You will need to listen to it with an analogue phone to see if it provides any inband signalling.
If your application needs reliable supervision, you need to use ISDN or use SIP, via an ITSP, who will, in turn, use ISDN.[/quote]
I doubt about it because in all other scenarios of calls there in no problem with disconnection/hang up. The problem only occurs when the call is passed to an internal number with the intervention of IVR.
In any case there is no built in IVR in Asterisk, so we would need to see your specific dialplan, as the IVR concept will be made up from lower level operations.
NB this board is not suitable for questions about dialplans created by FreePBX.
[quote=“david55”]I’ve already explained the reason for this.
In any case there is no built in IVR in Asterisk, so we would need to see your specific dialplan, as the IVR concept will be made up from lower level operations.
NB this board is not suitable for questions about dialplans created by FreePBX.[/quote]
Ok I understand what you mean now. Do you want me to upload some specific files?