We have a bunch of analog telephones and a few softphones and IP-phones in our office. Analog telephones run by a telephone exchange Samsung OfficeServ 7200 connected to a FreePBX. FreePBX has extensions for each analog telephone for call recording. When I try to call manually from analog telephone call is being recorded, all is fine. But when I use "channel originate" command from Asterisk CLI FreePBX does not record this call.
Command I use in Asterisk CLI looks like this:
channel originate DAHDI/i2/134 extension 8903XXXXXXX#from-internal
Where 134 is local analog telephone and 8903XXXXXXX is my mobile phone. What can I do to record originated calls as well?
You have 2 choices
1) Use Freepbx. If you use freepbx, you have generate calls via call Local/XXX#from-internal/n and set CALLERID acordinly. Recording will work just ok
2) Use custom code. If so, YOU are responsible for any logic like macro-record-enable or custom monitoring, so you have do additional dialplan to handle that.
Related
I want to program an esp8266 doorbell to call me when someone presses the bell. I have a STARFACE telephone system (Asterisk) and would like to tell STARFACE to make a broadcast call. I have searched the Internet but I find only FritzBox examples.
I do not want to do this with a call file.
Sorry for my English. I am not a native Englishman.
Call file is simplest way do that.
Some other ways
asterisk AMI protocol Originate command
asterisk ARI
perl,sipp(testing tool) or other script which send sip invite with auth.
https://gist.github.com/maximevalette/802764
http://sipp.sourceforge.net/
click2call script on asterisk (using call file or other)+ curl request on your device.
New to Asterisk here.
I've setup my phone as a gsm gateway using bluetooth adapter. I then direct calls from the mobile device to a queue that has one member. The member is a soft phone.
extensions.conf looks like:
[incoming-mobile]
exten => s,1,Answer()
same => n,Queue(support)
same => n,Hangup()
This works pretty well. I get the call to the sip soft phone. However, while the call is active, if I receive antother call to my phone device, that call deosn't even reach asterisk. I see the multiple call notification on my phone but it does not go through to asterisk. So my questions:
Is it even possible to queue concurrent calls via chan_mobile or does that kind of thing require more advanced hardware (e.g. gateways with multiple channels)
Even if I had some gsm gateway with 30 sim cards, what happens to the 31st call. How do people handle queuing the calls, once all channels are taken? Seems like it's certainly possible
Any pointers much appreciated
Some facts
1) chan_mobile not support any gsm gateway, it is clearly stated in project docs.
2) there are no gsm gates or channel drivers supporting second call on same gsm sim card.
3) There are no even PHONES which support that. On all phones you have pause/hold first caller to get second call.
Nobody need this feature, so no expect it will be developed in near future.
I want to send all extensions and conference bridge participants of asterisk voice to a analog telephone cable which is connected to a voicelogger ( recorder system) . How can it be done ?. I think this is possible by connecting Analog phone cable to ATA device ( linsys pap2) and sending stream to that ATA extension . But the challange is voicelogger is not an automatic answer machain .
First i have say you that idea is really strange. Asterisk can record all calls and record storage will cost much less then any analog device storage.
If you still insist you need it send to analog, you need multiple line analog device(every call record will require different wire).
Also you need FXS dahdi card and/or sip fxs adapter to connect your recorder.
You can orginize recording by using ChanSpy and/Or Confbridge as "ghost" call to all your calls with other dialling your fxs recording bank.
Complexity of such dialplan will be above average and require significant efforts and asterisk knowledge. You can read this links to get idea.
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Other options you can have is record by asterisk and play recorded files one-by-one to your analog recorder or just use usual computer to playback files to recorder.
I found an interesting documeent about realisation of eCall (Emergency Call) in EU: http://www.heero-pilot.eu/ressource/static/files/heero_wp3_d3-3_final-operational-results_v2.3_final.pdf
Germany somehow did it on Asterisk. Whatever, I don't understand how they process MSD (minimal set of data) using Asterisk. In the call session for the first step caller sends DTMF signals to send MSD packet. As I understood, Asterisk must redirect this call to In-band modem on COM port or to another machine with such modem. After PSAP successfully received MSD for the second step caller switches to voice channel that must be redirected to some sip-client of PSAP operator. How they do it? Is there a way to receive DTMF signals w/o modem by using internal capabilities of Asterisk? How the same call to redirect to another SIP on the same time?
I suspect that you are referring to emergency services, rendered by emergency dialers - eg. for senior citizens. These are fairly common where I live, and I've created in the past a solution to handle the calls from these, based on Asterisk. The solution involved a way to intercept the various DTMF signals that the device generates, then making Asterisk do stuff with it. Back then, I used Asterisk 1.6 and it is pain staking, because I had to do everything from within a MeetMe bridge, and interact with Manager alot. Today, doing the same with Asterisk 12/13 and ARI is a breeze. Just remember one thing, most of these dialers will utilize the A,B,C,D DTMF signals, which are somewhat unknown to most people - they exist and Asterisk is very much capable of handling those.
The only snag is - make sure you are connected via a PRI, as most SIP carriers aren't aware of these signals, and their SIP trunks won't support this type of signalling.
Asterisk can send dtmf(natively,via command SendDTMF in dialplan or D option in Dial command) or any other sound(custom c/c++ app needed)
No, you not need special modem.
However there are no realisation acordinly to that document, you need do that yourself or hire someone
I was wondering if it is possible to send via GSM protocol and appropriate AT commands, few data through a simple GSM call (witout using data traffic).
For example my system is a PIC + GSM Module and it monitors and processes a string of data.
When the user wishes, he makes a voice call in order to interacts with the GSM module via DTMF commands for example via an APP.
My dubt is if the GSM module is able to send the data to thesmartphone in order to monitor the status of the system. The module is a Quectel M95.
During some investigation, I saw that the module can send USSD codes but I don't know if it's possible to customize the USSD and read it from the smartphone as I wish.
Or maybe is it possible to use the FAX for data exchange?
Thanks everybody in advance
Marco
There are a couple of alternatives if you want to avoid using data traffic. But they all require either a connection to a service that is capable of receiving SMS's for example Twilio. Or you can of course have a GSM Module connected to your server which could do the receiving.
You could then send your data and requests for data within an standard SMS body. Or alternatively you could send binary SMS's where you are not limited to the 7 bit character set.
USSD would only work if you have a USSD service provider (I think Twilio has this now). Because the USSD service must initiate a USSD session for your target GSM Module to respond to. USSD code sending for GSM Modules is operator specific and therefore you are limited to what they have implemented (usually for prepaid users and the topping up of accounts).