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  1. Giga permalink

    Hello, I’m looking for a solution on asterisk to have and E1 gateway which will be working on SS7 signaling, do you have any tutorial on how to configure SS7 on asterisk?

  2. Georgie permalink

    Hi Sir, I am on mac, so how can i deal with it about Asterix following your courses.?

    Thanks, yours georgie

  3. Victor Adelegan permalink

    Hello Sir,

    Hope you are good!
    I’m contacting you cos I need to utilize the Asterisk to provide cheap calls to my immediate environs and to my country at large. I got to know about asterisk in my current workplace, but searching through the web I found your blog. Can you please be my mentor on this project? any help/guidance would be appreciated.

    Thank you.

  4. Michael Myers permalink

    I joined your class and I have a few questions. Can you please answer my e-mails please.


  5. Peyush permalink

    Hi Jonathan,
    First of all thank you for that great training video on asterisk. I have come from a non Linux background and you have explained it so well that anyone could understand it.

    We are looking into introduce asterisk as telephony solution and started the course.
    We have Skype for Business clients that we use as part of our O365 business premium license. Is it possible to integrate Asterik with our current SfB Clients ( O365 ) please ?


  6. khalid permalink

    Hello Dear

    I am new in Asterisk how I create the xml file for cisco ip phone 7945 i am using asterisk as call manager server

    Thanks in advanced

  7. Harvey permalink

    Looking for someone that can help setup an pbx extension on my iPad.
    I have a asterisk pbx systems and use sip.
    Thank you

  8. thulani permalink


    I would like to purchase your tutorial on voip, but udemy wont accept my payment. Please help.

  9. Walid MAgdy permalink

    Hi…I am very wonderful about your information in asterisk but i need to ask you where can i write this or in which file exten => 8888,1,Chanspy(‘all’|qb).
    Sorry but i am new in asterisk and need this functition.

  10. Ali Muharram permalink


    I have joined your course to learn Asterisk and have some questions. Can I follow your course using CentOS 7.x instead of 6.x? Also can this be done with Certified Asterisk 13 instead of 11? Thanks.

  11. Mark Tunde Adetayo permalink

    Hello Jhone Good day to you, I saw your video tutorial on Asterisk and want to say. grate job. but hope you have same for Freeswitch.

  12. Alexander Matyas permalink

    I have started hearing your Asterisk Training Udemy, but I am not sure if this relates to actual writing the call flow or its from the front End
    I have already created this flow from the front end and I would like to learn how to write it.
    If you have not included the writing lessons, would you be able to teach me to write in code. I would also want to learn to enable to write and read from/to a server

  13. Alex Pancorvo permalink

    I need to set a voip IPTABLES that only allows the incoming traffic and ports we authorized, we use Centos 5.9 and 6, is this something you can do?

    I was trying with the example somebody posted in yout blog but I only get errors.



  14. Rolando Azevedo permalink

    Hi Jon.
    I subscribed your Asterix course on Udemy, and learn a lot about Asterisk
    I have a question:
    How can i create an extension to automatically answer an incoming call and dependind on the selection forward the call to extension 100 or 101 for example.

  15. Devinder Sharma permalink

    Hi Jon, First of all, great work done by you and I am going to subscribe to few of your courses at Udemy. Just need some clarifications:

    1. Do you cover the GUI for Asterisk?
    2. Do you show how to use Cisco Phones (non-SPA phones) to enroll into Asterisk PBX and will BLFs etc will work on Cisco Phones? I had played with Asterisk little bit few years ago and I believe I could not make BLFs to work as that needed SIP TCP, whereas version I was using was 1.2 or something like that only supported UDP. Speed dials had worked though as also SIP trunking.
    3. Is there an easy way now to have a tftp server in the PBX to which we can upload required Cisco SIP image files and SEP XML config files so that Cisco phones can be auto enrolled or with little effort?
    4. Can soft SIP clients be reliably used while inside and outside the network on Smartphones so that extension can be extended thru Internet (wifi and cellular LTE etc.)?
    5. Are IM and presence functions available in Asterisk and shown in your lectures or if not, will you create another course for this?

    Thanks so much and great weekend.

  16. John Guarnieri permalink

    Hello Jon,
    Following your tutorial on “asterisk maid easy” and ran into an issue with make menuselect i get this error “Terminal must be at least 80 x 27” and it exits
    I am using centos 6.8 in a VM.

    Please advise.

    Thank You
    John Guarnieri

  17. John Guarnieri permalink

    tried to dial 8777275714 did not connect

    Connected to Asterisk certified/11.6-cert16 currently running on localhost (pid = 1199)
    == Using SIP RTP CoS mark 5
    — Executing [8777275714@mycontext:1] Dial(“SIP/101-00000003”, “SIP/18777275714@voipms”) in new stack
    == Using SIP RTP CoS mark 5
    [Mar 5 14:47:33] ERROR[1338][C-00000008]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(“voipms”, “(null)”, …): Name or service not known
    [Mar 5 14:47:33] WARNING[1338][C-00000008]: chan_sip.c:6113 create_addr: No such host: voipms
    [Mar 5 14:47:33] WARNING[1338][C-00000008]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 – Subscriber absent)
    == Everyone is busy/congested at this time (1:0/0/1)
    — Executing [8777275714@mycontext:2] Hangup(“SIP/101-00000003”, “”) in new stack
    == Spawn extension (mycontext, 8777275714, 2) exited non-zero on ‘SIP/101-00000003’

  18. Hugo Mendez permalink

    Hi Jon

    My first communiction with you!!, hope you’re doing well.

    I would like to know if you still on this matter, so I can address some questions to you.

    After that, with the 3CX company adquiring Elastix Solution, I would to know if asterisk is still reliable and we are able as enthusiastic user to work with it as freely as allways were.


  19. Brian Benson permalink


    I am enjoying your class on Udemy, thank you. Had a question for you, which I realize is well beyond the scope of the class, but was curious to pose it.

    I am setting up a FreePBX under Asterisk13, but wanted to understand the underpinnings better, which is why I took your class. We will be using PRI/Dahdi, so SIP phones won’t be an issue.

    But I was working on your course today, and I notice all these login attempts started happening to nonexistent extensions, like so:

    [Apr 5 11:12:23] NOTICE[1264]: chan_sip.c:27944 handle_request_register: Registration from ‘206 ‘ failed for ‘’ – Wrong password

    [Apr 5 11:12:33] NOTICE[1264]: chan_sip.c:27944 handle_request_register: Registration from ‘208 ‘ failed for ‘’ – Wrong password

    [Apr 5 11:12:35] NOTICE[1264]: chan_sip.c:27944 handle_request_register: Registration from ‘204 ‘ failed for ‘’ – Wrong password

    I emailed the abuse email of the ISP that shows up in the WHOIS for the above IP address. The IP is in Germany, I’m in NYC. I had used a very strong password fwiw in my test.

    So my question: is this normal, in other words, are they just constantly doing port scans and snooping for SIP installs in order to steal essentially long distance I guess?

    Thanks for any thoughts, sorry for the long explanation.


  20. Robert permalink

    Hi I had a question on one of your Udemy tutorials. Start your own voip business and quit your job in 90 days…..

    Because websites change and some of the content changes it is a bit difficult to follow you in this course when it gets to the part where you show how to install physdiskwrite and astlinux…..apparently the site in the video has changed and its a bit more challenging to follow. Although I do see that someone has attempted to address that part of the video.

    Will you be updating this anytime soon?
    I will send you the exact verbiage when I get home from work.

    If I could get past this hiccup I might be able to quit my job in 90 days.

    Im heavy into call centers….This course will really help me out.

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