Look at ${HANGUPCLAUSE}, you can determine if the reason the call was hung up was BUSY (or answered elsewhere etc). From there you can dial another exten if need be. www.asteriskdocs.org/
Hi .. Is it possible to listen the event from asterisk server like when I call to someone I need call details like caller,callee,start time,duration,end time etc.... if it possible please let me know I will learn
Actually non was given..I did however manage to setup outbound calls with no auth. New challenge: I need to initiate a call via nodejs ARI, when the callee picks up, I immediately put them on hold and make them listen to a radio stream configured in musiconhold.conf Progress: 1. I can make outbound calls 2. Stream icecast radio stream via music on hold What I cant figure out: How to immediately put the callee on hold when he picks up and make him listen to the stream. Thanks to your tutorial, I managed to setup outbound calls with no auth
That's odd regarding no auth, never seen that before. As to your problem (assuming predictive dialer) you would need to create a holding bridge. This may help: github.com/asterisk/ari-py/blob/master/examples/originate_example.py
excellent job....... i am working on the asterisk and sip trunk configuration but unfortunately iam not able to register with provider i don't know where i did wrong can u please help me .............
Hi I have 2 asterisk servers all configured. I tried connecting them together via PJSIP trunk but it doesn’t work because there is no option for me to enter pjsip details for incoming calls. I managed to get past this by allowing anonymous calls in but this is not a god practice. Would really appreciate any help 😀
Have never tried to be honest I always use iax (kind of what its for) and then send the direct the right dialplan to be able to reach extensions on both side or dial out from sip trunks on either side which are all pjsip/sip
Tried your config on both freebsd and debian and cannot make or receive calls. In your [from-sip] section - is that your sip number? All internal numbers and voicemail work and the sip is registered and showing as an endpoint.
having similar issue: res_pjsip_session.c:3962 new_invite: provider: Call (UDP:X.X.X.X:5060) to extension 's' rejected because extension not found in context 'from-sip'.
Could probably do one showing how to use AGI scripts, I don't use python but i can use bash, php, c etc and the language behind it us less important than how variables are passed to the scriot and back, such as a php script i use ti log every call ti slack. Or the one we use which posts answered call count per employee and missed call count at the end of each day
good afternoon sir, I am a systems student, please I would like to know if it is possible to implement an asterisk server and make external calls (to cell phones) simply by having an internet connection? And if not, what should I take into account? Thank you.
If calls are only to be made via an internet connection, you'd need a voip provider to provide you with siptrunks, you pass the call to the sip provider and they handle cellular/pstn routing
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Thanks a lot!
Now my outboud calls are working!
Glad it helped, thanks for feedback
great videos! asterisk always was a complicated thing to me, now I see that it could be simple.
It looks daunting at first, pretty simple when you move most of the configs out of the way and scratch.
Thanks for the feedback!
Thank you so much that helps, was trying to use the wizard but the endpoints will not connect. Wierd.
Glad it helped 🙂
Hello sir I need to know how to forward call if the extension is busy thanks
Look at ${HANGUPCLAUSE}, you can determine if the reason the call was hung up was BUSY (or answered elsewhere etc). From there you can dial another exten if need be. www.asteriskdocs.org/
I can call my friend ( not same network) like call normal FAI?
You'll require a sip trink provider to handle external calls
Hi ..
Is it possible to listen the event from asterisk server like when I call to someone I need call details like caller,callee,start time,duration,end time etc.... if it possible please let me know I will learn
Yes you can using various libraries, Asterisk Manager (built-in) allows you to do this, for example sending incoming call notifications to slack
@@sheridans can u share any video reference or I will provide my contact details can you teach me
Hiring us has a cost, let me see if I can sort a video on this...
@@sheridans ok pls help me I need this
Hi, good tutorial.
my provider only provided an IP. How do I make an outbound call without authentication
This doesn't make sense, they should've provided you with credentials to authenticate.
Actually non was given..I did however manage to setup outbound calls with no auth.
New challenge:
I need to initiate a call via nodejs ARI, when the callee picks up, I immediately put them on hold and make them listen to a radio stream configured in musiconhold.conf
Progress:
1. I can make outbound calls
2. Stream icecast radio stream via music on hold
What I cant figure out:
How to immediately put the callee on hold when he picks up and make him listen to the stream.
Thanks to your tutorial, I managed to setup outbound calls with no auth
That's odd regarding no auth, never seen that before.
As to your problem (assuming predictive dialer) you would need to create a holding bridge.
This may help:
github.com/asterisk/ari-py/blob/master/examples/originate_example.py
would have been nice if you had full paged putty, hard to follow in that small window!!
Thanks, point taken and noted.
excellent job....... i am working on the asterisk and sip trunk configuration but unfortunately iam not able to register with provider i don't know where i did wrong can u please help me .............
nothing shows up in the console with asterisk -rcv? can also use core sef vervose and core set debug commands for extra output
Hello friend, I need to know how I can, after calling an external number, forward to another internal extension to listen to a recording. Thank you.
I am just doing videos on project's currently undertaking. Asterisk has great community support community.asterisk.org/
If you post on our forum we can take further
Hi Rafael, did you figure this out yet?
Hi I have 2 asterisk servers all configured. I tried connecting them together via PJSIP trunk but it doesn’t work because there is no option for me to enter pjsip details for incoming calls. I managed to get past this by allowing anonymous calls in but this is not a god practice. Would really appreciate any help 😀
Hi, I'd recommend connect them together via iax2, that's how I usually set them up
@@sheridans thanks for the fast reply. I have managed to connect them via IAx and SIp was wondering if it can be done via pjsip
Have never tried to be honest I always use iax (kind of what its for) and then send the direct the right dialplan to be able to reach extensions on both side or dial out from sip trunks on either side which are all pjsip/sip
Tried your config on both freebsd and debian and cannot make or receive calls. In your [from-sip] section - is that your sip number? All internal numbers and voicemail work and the sip is registered and showing as an endpoint.
having similar issue:
res_pjsip_session.c:3962 new_invite: provider: Call (UDP:X.X.X.X:5060) to extension 's' rejected because extension not found in context 'from-sip'.
Hi, can you make tutorial on how to use custom AGI scripts in asterisk (preferably in Python)
Could probably do one showing how to use AGI scripts, I don't use python but i can use bash, php, c etc and the language behind it us less important than how variables are passed to the scriot and back, such as a php script i use ti log every call ti slack.
Or the one we use which posts answered call count per employee and missed call count at the end of each day
good afternoon sir, I am a systems student, please I would like to know if it is possible to implement an asterisk server and make external calls (to cell phones) simply by having an internet connection? And if not, what should I take into account? Thank you.
If calls are only to be made via an internet connection, you'd need a voip provider to provide you with siptrunks, you pass the call to the sip provider and they handle cellular/pstn routing
@@sheridans please I want your email..