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.
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
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.
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.
I'm making an outdial using call files in asterisk and application needs some DTMF input but DTMF not working for all mobile phones, It is not accepting digits from Nokia - 1100 and nokia 6030 where DTMF works if i make an incoming callfrom the same phone but on out dial using DTMF log i can see following messsage on asterisk CLI:
DTMF Begin '1' received on Dahdi-32
DTMF Begin ignored '1' on Dahdi-32
If someone knows the solution for this please let me know.
It works if i set overlapdial=no but in this case it doesn't make two simultaneous outdial on two different numbers.
Have you tried setting relaxdtmf=true in the chan_dahdi.conf file?