I am new on Asterisk.
I have configured and installed Asterisk + FreePbx with 4 USB MODEM as FXO.
All is working fine, but I need block or hangup all inbound calls.
I possible make it from dondle.conf ? or where?
Thanks.
Do do this you need to edit the context in the dongle.conf and then create a context in extensions.conf
The extensions.conf context can read
[dongle-inbound]
s,1,Hangup()
That will simply hangup all calls that are passed to that context.
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.
I am trying to dial a call to GOIP and want to forward it to another GSM phone. I have added following lines in my extensions.conf (asterisk)
[goip_context]
exten=> 105,1,Set(NUMBER=913579865476)
exten=> 105,2,Dial(SIP/${NUMBER}#${GW2},20)
where 105 is my GOIP extension
NUMBER is the dialled number
and GW2=goip
On dialing a call, following logs appear on asterisk terminal
Call to peer 'goip' rejected due to usage limit of 1
-- Couldn't call SIP/913579865476#goip
Please guide in this regard
Acordinly to message you have call-limit in sip.conf set to 1 for this section
Also you need at least 2 gsm modules to do that.
We have a soft phone that's dialing out, on a SIP trunk, through our Asterisk server. The soft phone is sending X Headers that we want to send on to the destination.
We see the headers coming into Asterisk, but not going out. Is there something we can do to forward the headers along to the destination?
Asterisk is no SIP proxy but a B2BUA. This actually means, that it is not forwarding the original request. The call from your softphone gets terminated on Asterisk. Asterisk starts a second call to the logical destination of your call and connects both calls together.
This is why on the outgoing call you don't have the headers you added on the incoming call. The outgoing call is initiated by Asterisk. If you want to add any headers from the incoming call, you manually have to to it like miken32 said:
exten => _X.,n,SIPAddHeader(X-Foobar: ${SIP_HEADER(X-Foobar)})
exten => _X.,n,Dial(SIP/${EXTEN})
Just tell your dialplan to pass them on; maybe something like this:
exten => _X.,n,SIPAddHeader(X-Foobar: ${SIP_HEADER(X-Foobar)})
exten => _X.,n,Dial(SIP/${EXTEN})
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 need to create a ring group (222) which would dial several SIP accounts, and PSTN numbers as well.
For PSTN I have a different context (ToPSTN) with it's own billing rules, so the question is:
How can I ring several SIP acc's and PSTN's simultaneously ?
Here is how I am trying to do that:
exten => 222,1,Dial(SIP/ca-444&SIP/ca-433)
exten => 433,1,Goto(ToPSTN,0035853855453,1)
Or maybe it's possible to do several tasks at the same priority somehow ?
To make dialing into dialplan instead of real channel driver you should use Local channel. This is how it look in your case:
exten => 222,1,Dial(SIP/ca-444&Local/0035853855453#ToPSTN)
So first call goes to SIP peer ca-444 and second directly to dialplan extension 0035853855453 and context ToPSTN.