Is there any way that i can forward a call from my landline number to my DID number and handle that call with asterisk code??
Is this type of forwarding possible?
There are 2 posible variants
1) Ask your landline provider support about number portability(depend of country and provider) or forward number to sip.
2) Put on your line FXO gateway device(pstn2sip gate, $100-$200 cost) and setup it to work as did for asterisk.
Related
I am working with an asterisk software pbx.
I have an IP phone which is configured with asterisk.
What i want to do is make call to a special number when the user hangs the phone. I do not want him to have to compose a number.
Do you know if it possible ?
Thanks
You have call, part A(caller) call to party B(called).
So.
You can setup your dialplan if B hangup, asterisk connect A with next number.
You can't setup your dialplan if A hangup, B connected to other number, except variant when you do connect A&B via conference(not via Dial command).
If you want phone call B when A get it, that called HOTLINE and it is feature of phone, not asterisk.
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 have VoIP infrastructure in company,
end points can dial mobile numbers in this case:
they call 9... they wait for pstn dial ton...after dial-ton they can dial their number.
i do this in this way:
exten => 9,1,dial(sip/8003)
witch sip/8003 is a sip account that is connected to FXO gateway and connected to asterisk via sip trunk.
i want to do this:
the end points dial:
909121111111
instead of
9... after-dialton.... 09121111111
Without having seen more of your dial plan, typically what you can do is, in an accessible context, a way to dial the whole thing -- and then use a substring to strip out parts of it.
exten => _9XXXXXXXXXX,1,Dial(SIP/8003/${EXTEN:1})
The first part of the extension matches a 10-digit number prefixed with a 9: _9XXXXXXXXXX. Check out the article on pattern matching on the Asterisk wiki
Next, on our dial application, what we do is dial your sip device, but, we pass it back the dialed extension, but, notice the colon? Like ${EXTEN:1} That's to strip digits. Namely it strips the first digit. You can learn more about manipulating variables on the wiki, too.
As the title states this is about GSM call forwarding.
When a mail is forwarded it's headers shows that it's forwarded. Is it the same when forwarding a gsm phone call?
More specifically if I set my phone to forward calls, in case I don't answer, to an asterisk server. Will I on that server have both the phone numbers? That is both the original caller and the one forwarding?
Is your asterisk box using DSS1? DSS1 allows you to get the redirecting number (the GSM subscriber number in your case).
I am new to Asterisk. I am not clear about my concept and don't know is it possible one.
Mobile number divert all incoming call to Virtual Telephone Number. Virtual number divert to Asterisk PBX using SIP. If Asterisk receive a call, is it any possibility to get the phone number from and original destination number in Asterisk
Thanks
Check the sip headers and see if the itsp is passing it