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 âbog standardâ 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 notâŚ?
Does anyone know if FreePBX supports this and if so, how to implement it?
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.
Not to rain on anyoneâs parade but all these responses are wrong. the âbog standardâ FXO will ring until it times out, then will bridge the call to itâs local voicemail at this point the channel is still available and a âbog standardâ FXS can still âflashâ the copper pair line and connect to it as can any asterisk channel technology , so for this to work, you canât have asterisk âanswerâ the channel as it is a service prior to answer. if you donât 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 âflashâ 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 chan-dahdi.so and Asterisk dialplans work, so not easy or cheap
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.
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.
This is shame. It is disappointing that the system is unable to do this âout of the boxâ . 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 iâll 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 âbarge inâ - 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 youâre looking for.
That said, an âaudio call screeningâ 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 http://issues.freepbx.org/. If there is enough interest, it may be considered.
Yes, understand. Iâm 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 altogetherâŚ
A possible âsynthesisâ 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 but
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?
would you get most of the way, but it doesnât encompass the r (record) option, a quick look shows it was written for meetme not confbridge (asteriskâs prior to v 10) a lot has changed since then but it âshouldâ work
To reiterate the important part of what @dicko said - this isnât 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. Itâs a very ânicheâ application that most businesses wouldnât actually want, but for home users, it might be interesting.
Thanks! I donât 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 donât âbreakâ 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 âout the boxâ
Because of the way that weâve âdesigned (if you can use the word in this context)â it, it should work with anything with an extension number and can do some kind of automatic âoff hook and play intercomâ. Itâs going to get weird in the middle, and you might end up not liking the system much once you get to that point, but thereâs no reason why you shouldnât eventually be able to make something like what youâre 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 wonât 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.
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 âthat doesnât sound very spy-ishâŚâ
The idea of ChanSpy is for supervisors to listen to CSR calls and help with problems that come up. Itâs not really intended to be done automatically, although (like just about everything else the system does) it could probably be programmed to do that.
This is something you can do within Sangomaâs Switchvox product and with the D-Series telephones attached to Switchvox. There, you can listen to a voicemail as itâs being left, and if you decide the call is important, you can pick up the call.