Listening to voicemail in real time as a user is leaving it


I have just deployed a FreePBX installation consisting of a handfull of Sangoma S500 and S700 phones at my parents house here in the UK. I also have a FreePBX deployment at my company office (also using Samgoma S500 phones).

The internet connection out in the countryside where I am is terrible, so I am still utilising the analogue BT line; this feeding a Sangoma A200 card installed in the PBX.

One of the main features that they require is the ability to listen in real time to a caller leaving a voicemail message and if it is a caller they recognise, pick up a handset on any of the phones to interrupt the answering machine. In other words, just like their previous “bog standard” analogue telephones! However, I cannot seem to find any settings related to doing this and Google is not much help either. I have thought this would be a much needed request, but it seems not…?

Does anyone know if FreePBX supports this and if so, how to implement it?

Many thanks,


My best guess would be no, reason being:

The phones are making a SIP connection back to the PBXServer. When the phone is ringing the server is sending SIP requests to the phone, once voicemail kicks in that is nothing but a software intercepting the call and its being recorded on the server side not the phone side.

Once the voicemail has been received the server notifies the phone that it exists. So without some other type of hardware intercepting the call after the PBX or directly registering the phone to a SIP provider with no PBX i dont think its possible. Maybe using an Analog line may work… not sure however.

Nope - same reasoning as your original point. It doesn’t make any difference what the telephone’s technology is.

The voicemail is being processed in the PBX. The phone is no longer part of the call, so there’s no way to insert yourself back into call stream.

Not to rain on anyone’s parade but all these responses are wrong. the “bog standard” FXO will ring until it times out, then will bridge the call to it’s local voicemail at this point the channel is still available and a “bog standard” FXS can still “flash” the copper pair line and connect to it as can any asterisk channel technology , so for this to work, you can’t have asterisk “answer” the channel as it is a service prior to answer. if you don’t then the call is normally not channel associative, so you would have to set a global variable , maybe __dahdi-chan-1 or some such and then exit leaving the channel open but unbridged and system identified, then theoretically your sip phones could subscribe to that global hint you just made, the bigger problem is getting the audio from the unanswered dahdi channel,maybe you could multicast it so all your extensions could listen, to answer the channel however you would need to “flash” that channel from the eventually answering listening extension and thus answer it . then the external voicemail bridge is terminated and you would have a standard dahdi analog call to process.

It is an interesting situation and I think quite resolvable, but it would take a lot of custom programming and someone who really understands how and Asterisk dialplans work, so not easy or cheap :wink:

Dicko not sure how I am wrong here. You cant do this with a SIP line. You are mentioning FXS ports which I did say… Maybe using an Analog line/Phone may work. He was trying to use Sangoma SIP phones. The phones are not programmed to take voicemail locally, they are meant to have them on the server.

Yes maybe with tons of re-coding and the right phone he may be able to get it done, but the OP would be much better off buying a regular home phone with a voicemail system for the $1000+ cost in redesigning an entire voicemail system and phone. :slight_smile:

A How about having an analogue traditional answering machine to take the message rather than voicemail then they can either pickup that hand set or Add the custom dial plan entries to do a barge in from another handset.


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Hi all,

Many thanks for the replies.

This is shame. It is disappointing that the system is unable to do this “out of the box” . i would have thought that this would have been a requested feature for small business and home users (who have large homes)?

It looks like i’ll have to create an analogue extension with an ordinary answerphone hooked up to it, although this does seem somewhat of a backwards step… and the user would have to be in ear-shot of the device to listen to the message come through. With regards to the “barge in” - it is possible to setup FreePBX to interrupt an extension once it has answered (which is what the analogue answerphone will do)? Then there is the problem of shutting the answerphone off and not having it record hours of conversations unnecessarily.

Maybe I need to look at a different system? 3CX, Cisco, Pamasonic etc?



No single system can be all things to all people. What appears to be a critical feature for you has come up in the forum exactly once (to my memory) and I doubt it has ever come up in the issue tracker as a feature request. At the end of the day, call handling features are largely limited to what can be done with Asterisk, and it is is only in recent versions (with the addition of the ARI) that the necessary framework has become available to do the types of things like the feature you’re looking for.

That said, an ‘audio call screening’ feature sounds like something that might resonate with the user community, so if you feel strongly about it, by all means file a feature request at If there is enough interest, it may be considered.

Yes, understand. I’m just surprised that this has not been a more requested feature. However, I am relatively new to VoIP and have not carried out any large installations.

Before resorting to a POTS answerphone, might there be a way of using an IP SIP PA speaker setup as an extension to auto-answer after so many rings - with FreePBX set to automatically record all calls? However, getting these recordings into the voicemail chain might be another task altogether…


A possible ‘synthesis’ if your phones support auto-answering,

Answer the call,
Play an announcement that sounds like a vm greeting.
Send the call to a full duplex page() group wirh recording enabled
If the caller is answered then ok
If the caller hangs up or the page times out, close the recording and add it to one or more vm boxes (a little complicated there because of the way comedian mail works :wink: but

has a bunch of bashism’s that might help


Or juat email it.

err ok. No nothing about setting up a full duplex page group. Could you point me in the right direction please? Is this something that can be done entirely in FreePBX, or will I need to use 3rd party tools as well?

Many thanks,


Np , probably the gui wont help here, just need a basic custom context and a lot oF RTFM :wink:



would you get most of the way, but it doesn’t encompass the r (record) option, a quick look shows it was written for meetme not confbridge (asterisk’s prior to v 10) a lot has changed since then but it ‘should’ work :slight_smile:

To reiterate the important part of what @dicko said - this isn’t something you can do from the GUI. You are going to need to do some programming if you really want this to work. Using the conferencing app will get you a good simulation of what you are looking for, but the outcome of the process may end up being messy.

If you decide to pursue this, let us know as you move forward. It’s a very “niche” application that most businesses wouldn’t actually want, but for home users, it might be interesting.

Thanks! I don’t mind getting stuck in with some programming, as long as I can have some guidance along the way. Can always setup a test machine so I don’t “break” the system running the house.

I take it then that we agree the hardware (Samgoma phones + FreePBX) is capable of doing what I want. Just not “out the box” :wink:



Because of the way that we’ve “designed (if you can use the word in this context)” it, it should work with anything with an extension number and can do some kind of automatic “off hook and play intercom”. It’s going to get weird in the middle, and you might end up not liking the system much once you get to that point, but there’s no reason why you shouldn’t eventually be able to make something like what you’re trying to do.

FreePBX can easily be programmed to do this using the Channel Spy application. You could listen in on a call and implement a touchtone button that would transfer the call out of voicemail and back to your extension.

I have a very small system and use the Channel Spy feature code all the time to listen in on the VM as it is being left. Just using the Channel Spy feature code won’t let you force a transfer back out of VM, but it demonstrates the easy with which you can listen to the VM while it is being left.

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Thanks, this sounds very interesting and promising. Can the Channel Spy application be programmed so as to automatically open up the loudspeakers on all extensions to play the voicemail, then from whatever extension you are beside, you then press a touchtone button to transfer the call out of the voicemail?


My first reaction was “that doesn’t sound very spy-ish…” :slight_smile:

The idea of ChanSpy is for supervisors to listen to CSR calls and help with problems that come up. It’s not really intended to be done automatically, although (like just about everything else the system does) it could probably be programmed to do that.

If there was a feature request for this, I would vote for it.

Can someone link if there is? I did not see one.

This is something you can do within Sangoma’s Switchvox product and with the D-Series telephones attached to Switchvox. There, you can listen to a voicemail as it’s being left, and if you decide the call is important, you can pick up the call.