Latest Asterisk

Asterisk 1.8.14.0 has been released a few days ago, so I’ve created a new image including this release as well as the latest fixes and improvements. You can find it on the downloads page.

25 thoughts on “Latest Asterisk

  1. I would like to express my appreciation for your great effort. I have your latest working well with GoogleTalk using the OBi interface. What a dream, all in a credit card size package. I had your previous image but could not get the CDR reports working. Your latest image took care of the fix. All is working just great for my ten phone system. Again thanks for making it possible.

    Gary

  2. Thank you for making this happen. I have been using freepbx and asterisk for nearly two years and due to relocating, I thought I would have to shelve this hobby/ communication method and leave the server behind. Now thanks to you, I’ll be able to set a pi of mine with asterisk and still have my desk IP phone. Thank you.

  3. Gernot, did you do any optimizations to debian or asterisk? I set up asterisk myself on my raspberry (using debian wheezy), but sometimes I have short moment of complete silence (half second or so) during my calls, which is not the case when using my SIP provider directly, without my raspberry. Thanks for any hint!

    • No I have not done any optimizations so far. I am also not yet using the new Raspbian with hard float support, this is still to come. The image posted on the downloads page uses the old Debian 6, and I have not experienced any silence or dropped calls so far. And I’ve been using the RPi myself for many >20min calls already.

      • Hello,

        First of all, Many thanks for your very great job.

        Have you planned a new image with raspbian soon ?

        Thanks

        Mickael

        • Yes I have planned a new image based on Raspbian, but it will take a few more weeks.

          • good to hear that, thank you for ur great work. I try ur instruction in wheezy but in result error 500 when login to freepbx… better wait your distro…

    • No I haven’t considered this. Stability of the overall system is a major concern for me, so I would rather rely on a base system that is well tested by other RPi users.

  4. Hi,
    first of all, thanks for this great implementation
    second, excuse my english.

    I´m planning a project to use my raspberry as a house phone system.
    By the way, i never collect skills in asterisk/freepbx before and never heard about it since two days ago, but i love it to tinkering in linux and electronics. :)
    So i took the latest image from http://www.raspberry-asterisk.org, config my static ip, etc… read a lot about asterisk and freepbx,… and added first sip-devices (android-smartphone and pc) over freepbx. I tested it with voicemail etc. and it works very well.

    But by doing the next step, i get grey hairs.
    The theme is “console dial”. I find out, that it is possible to use linux alsa sounddevices as a softphone that can control with asterisk console by typing e.g.”CLI>dial xxx” or “CLI>console dial xxx”
    I need this function to couple the mic/speaker from raspberry to the speaker/mic from an exsisting door-phone-installation. (Also the gpio should used by shell script to control the electric door opener. I will do this later, because i need first to proof the “console dail” concept and also control gpio is not so difficult.)

    So what have i done:

    ————————

    getting the asterisk-console:

    -install many admin-modules at freepbx, also the module “Asterisk CLI”

    -connect via shell and execute “asterisk -c” (there are failure-text at start, but i get a console “CLI>” I think the errors are because of starting as user “pi” and not as linux-user “asterisk”, so asterisk runs parallel at the same time!?)

    ————————

    soundcard (usb):

    -install a (alsa) usb soundcard

    -type “cat /proc/asound/cards” give me only the installed usb device:
    0 [Device ]: USB-Audio – USB PnP Sound Device
    C-Media Electronics Inc. USB PnP Sound Device at usb-bcm2708_usb-1.2, full spee

    -i can record in shell: “rec ~/test.wav”

    -i can play and hear it: “play ~/test.wav”

    ———————-

    asterisk config:

    -edit the /etc/asterisk/modules.conf and change “chan_alsa.so” to “load”
    (chan_oss.so is by default noload)

    -edit the /etc/asterisk/alsa.conf in many different configurations
    e.g. like this how-to http://www.voip-info.org/wiki/view/Asterisk+CLI+dial
    (autoanswer=yes, context=local, extension=s, mohinterpret=default, input_device=hw:0,2, output_device=hw:0,1 etc…)

    ———————-

    But nothing worked!

    When i type “dail 2000″ or “console dial 2000″ at CLI> to call e.g. my pc using, i get always the same error on bash and on asterisk-cli-module:

    “No such command ‘dial 2000′ (type ‘core show help dial 2000′ for other possible commands)”

    I found out, that the dial command only available if the alsa or oss conection to asterisk is existing and working.
    So, how can i proof this or what is wrong in my config?
    Is there a way to configure console dial by using only freepbx?

    It would be nice if someone can give me a hint.

    Thanks

  5. did anybody managed to install chan_dongle for huawei voice datacard modems (usb) in asterisk on raspberry Pi?

    I’ve got Huawei E173 installed, but getting errors like:
    [2012-07-31 12:07:07] ERROR[10332]: chan_dongle.c:433 do_monitor_phone: [dongle0] timedout while waiting ‘OK’ in response to ‘AT’
    — [dongle0] Error initializing Dongle
    — [dongle0] Dongle has disconnected
    — [dongle0] Trying to connect on /dev/ttyUSB2…
    — [dongle0] Dongle has connected, initializing…

    any help :>?

      • Gary, first of all I would like to say thank you for this wonderful piece of work.

        Regarding the root password, i had some problems using the default pi user. More specifically, i was trying to upload a file using sftp, and the default pi user didnt have the correct perimissions. For users who are getting errors when using the pi user, they can try activating the password for the root user.

  6. Hi, I am not able to connect by a extension. I have created extension 100 and the secret key. I have a app on my iphone and when trying to connect it just times out. In the app I am putting the username as 100 and password the secret key. Is there anything I am missing?. I have checked the logs freepbx.log but nothing useful shows up it’s like the registration is not even reaching the server. Any suggestions?

      • When using the image for the first time for SIP, you need to properly configure Asterisk SIP settings from the FreePBX menu, and then restart Asterisk with amportal restart (reboot will do as well of course). Otherwise the SIP port 5060 is not activated. Maybe this is a bug in FreePBX, as pressing the “Apply Config” button is not enough to enable the SIP port.

      • Just applied chgane to the conf file. ( user = root )It seems to be working.However, remind the folks out there that if they are using Flashybrid , to save their chganes with fh-sync or they will lose their chganes on the next re-boot.

  7. Hello,

    I’m from spain and i’m studying vozip in the college. I have installed your image, but i have had some errors:

    Sorry for my English.

    1.Need to update the time:
    Install ntupdate
    sudo apt-get install ntpdate
    sudo apt-get update
    and reboot

    2.Configure the interfaces static.
    3. Purge and reinstall SSH.
    4. Generate user in the file /etc/asterisk/sip.conf
    5. To make call via NAT, i have to fordward the port 5060 to RPI, and also 10000 to 20000.

    After i can change and create the user throug FreePBX.

    Thank you so much for your time, i am using your image to do practice with asterisk.

  8. Gernot, may I ask what the RAM usage of your images (when running, including Asterisk) is? I did my own setup and feel like this is a problem in my case.

    • I did not see more than 120 MB used on my system so far:

      pi@raspberrypi:~$ free -m
                   total       used       free     shared    buffers     cached
      Mem:           218        196         21          0         24         52
      -/+ buffers/cache:        119         98
      Swap:          196          0        196
      
  9. Does anyone know the path and filename of the startup script run that activates the whole “requires a time server to start” thing? I need to make some modifications to that. Thanks

    • You are right, this script needs some improvement. You can find it in:
      /etc/init.d/freepbx

  10. first Thanks for this work … it looks promessfull!!!

    But I’ve got a question …
    My Raspberry is working with your configuration …
    Got the Web interface Ok.
    Connect on the asetrisk interface with asterisk -rvvvv ok
    Declaring and registred Registred SIP Phone ….
    No Sip line to outside for now ….

    So that’s nearly Good ! ….

    Except that i’cant even call my other phones ….
    it says : app_dial.c:2341 dial_exec_full : unable to create channel of type SIP (cause 20 – unknown)
    and lot of warning a ipbx reload …

    Any Idea ???