Il 20/04/2017 17:32, kevin.larsen at pioneerballoon.com ha scritto:> > This gets kinda Rube Golberg-ish, but convert the incoming analog line > to sip, route it through asterisk and have asterisk do its thing > before converting it back to analog to send to the phone. Only problem > is you get a lot of extra hardware involved in the mix to make it > work. It will be a lot of expense and trouble, so you need to make > sure that whatever part you want asterisk to play is worth that > effort. Also, I wouldn't touch a fax line in this manner. > > If you could give a bit more info on what you want asterisk to do, we > could maybe give better advice on how to solve your problem.Hi Kevin, I've already proposed your solution (is the most reasonable) but they have more than 60 analogs lines (no faxes) and some of them terminate in appliances like alarms, etc, so the solution must not touch in any way the connection between the line and his termination: doing a analog to digital conversion, passing it to asterisk and the convert it back to analog is prone to problems (what if asterisk crashes? or if a gateway fail?). I can split the existing lines (there are no complex things like adsl or digital signaling), convert the branches to digital and terminate then into an asterisk machine, so any failure will not affect the old circuit, but of course I've to configure asterisk to ONLY LOG calls and nothing more. This is what they want: - line 1 ring - line 1 is splitted in two, the first branch (let's say the "analog" branch) go to an analog phone, that rings - the second branch go through a gateway and then to asterisk - asterisk log (with an AGI for example) "line 1 rings at .... from ...." no more is required from asterisk, if someone answer the analog phone or not is not my business.
kevin.larsen at pioneerballoon.com
2017-Apr-20 21:09 UTC
[asterisk-users] log incoming calls without answering
> I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to > digital conversion, passing it to asterisk and the convert it back to > analog is prone to problems (what if asterisk crashes? or if a gateway > fail?). > I can split the existing lines (there are no complex things like adsl or > digital signaling), convert the branches to digital and terminate then > into an asterisk machine, so any failure will not affect the old > circuit, but of course I've to configure asterisk to ONLY LOG calls and > nothing more. > > This is what they want: > - line 1 ring > - line 1 is splitted in two, the first branch (let's say the "analog" > branch) go to an analog phone, that rings > - the second branch go through a gateway and then to asterisk > - asterisk log (with an AGI for example) "line 1 rings at .... from...."> no more is required from asterisk, if someone answer the analog phone or > not is not my business. >Ok, so I would agree with them that a conversion to digital and back again would tend to break things like fax lines and alarm lines. My analog lines in my facilities are there because a lot of alarm systems just don't work with SIP at all. It's something the alarm companies are going to have to figure out in the next decade or so as the Telcos are moving away from copper and switched networks and towards fiber and packet based networks. I honestly don't know if you can do what you want without some piece of equipment picking up the line. What I would do is get an analog line, an analog phone, an analog to sip device (there are many to choose from) and a basic asterisk instance. I would then make a small test setup where the analog line goes to a splitter. One side of the splitter goes to your analog phone. One side goes to your analog to SIP converter and then into your asterisk instance via your ethernet network. Use your cell phone to call the number of your analog line and see if it works. You would have to code a basic dialplan on the asterisk side and set up the trunk to your converter, which I am assuming you know how to do. This would at least give you a fairly low cost way to test to see if you can trigger what you want on the Asterisk side without also triggering the line itself to be answered. I would also note that you would only be able to log incoming calls this way. I can't see a way you would be able to detect an outgoing call from the analog extension. ______________________________________________________________________ This email has been scanned by the Symantec Email Security.cloud service. For more information please visit http://www.symanteccloud.com ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170420/8e964e57/attachment.html>
Il 20/04/2017 18:09, kevin.larsen at pioneerballoon.com ha scritto:> > I honestly don't know if you can do what you want without some piece > of equipment picking up the line. What I would do is get an analog > line, an analog phone, an analog to sip device (there are many to > choose from) and a basic asterisk instance. I would then make a small > test setup where the analog line goes to a splitter. One side of the > splitter goes to your analog phone. One side goes to your analog to > SIP converter and then into your asterisk instance via your ethernet > network. Use your cell phone to call the number of your analog line > and see if it works. You would have to code a basic dialplan on the > asterisk side and set up the trunk to your converter, which I am > assuming you know how to do.Yes, I'll definitely do the test before set up the whole proyect, but the point basically is: it is possibile for asterisk to log a call without answering it? How to do it in the dialplan? Or I'm wasting time because an analog line who enter asterisk is always answered?
On Thu, Apr 20, 2017 at 05:51:59PM -0300, Fabio Moretti wrote:> Il 20/04/2017 17:32, kevin.larsen at pioneerballoon.com ha scritto: > > > > This gets kinda Rube Golberg-ish, but convert the incoming analog line > > to sip, route it through asterisk and have asterisk do its thing > > before converting it back to analog to send to the phone. Only problem > > is you get a lot of extra hardware involved in the mix to make it > > work. It will be a lot of expense and trouble, so you need to make > > sure that whatever part you want asterisk to play is worth that > > effort. Also, I wouldn't touch a fax line in this manner. > > > > If you could give a bit more info on what you want asterisk to do, we > > could maybe give better advice on how to solve your problem. > > Hi Kevin, > > I've already proposed your solution (is the most reasonable) but they > have more than 60 analogs lines (no faxes) and some of them terminate in > appliances like alarms, etc, so the solution must not touch in any way > the connection between the line and his termination: doing a analog to > digital conversion, passing it to asterisk and the convert it back to > analog is prone to problems (what if asterisk crashes? or if a gateway > fail?). > I can split the existing lines (there are no complex things like adsl or > digital signaling), convert the branches to digital and terminate then > into an asterisk machine, so any failure will not affect the old > circuit, but of course I've to configure asterisk to ONLY LOG calls and > nothing more. > > This is what they want: > - line 1 ring > - line 1 is splitted in two, the first branch (let's say the "analog" > branch) go to an analog phone, that rings > - the second branch go through a gateway and then to asterisk > - asterisk log (with an AGI for example) "line 1 rings at .... from ...."Simple dialplan. Depending on the type of caller ID system, you may need to wait a few seconds (in case the caller ID is sent after the first ring). Thus, assuming you have a DAHDI device, your dialplan is: exten => s,1,Wait(5) ; check how much and if waiting is needed same => s,n,NoOp(Caller ID is ${CALLERID(num)} on DAHDI channel ${CHANNEL(dahdi_channel)}) And move on to report from there. If you also need to report the total time of the call: that might be possible if the remote side reverses polarity of the channels on call start and end. Information about it is currently only reported in debug messages by chan_dahdi. So it is possible (given polarity reversal), but tricky. -- Tzafrir Cohen +972-50-7952406 mailto:tzafrir.cohen at xorcom.com http://www.xorcom.com