Great tutorial, just compiled Asterisk-18.3. Few changes but this led me through the whole process nice and easy! Thanks very much for this great series
Conrad - thanks for the shoutout and for picking up my Suggestion. I have tried several times to Setup Asterisk and I often failed. Your Videos took me up to the next Level. My very simple RPI-PBX is already running. I will now take an Espresso and watch this Video. All you Guys out there - Take Care and Stay Save! Greetinz from Germany, Gerd
@@InnovateAsterisk Ok - here we go: I installed Asterisk 16, following your Tutorial and everything went fine. There is one Problem at least. Asterisk does not load the codec_opus_arm.so … It is available in the "/usr/lib64/modules/asterisk" -Folder but a "core Show Translation" does not list the Opus Codec. Do you have a Suggestion, what the Problem could be? A "core show codecs" lists the Opus Code.
Yes, this the same as what I experienced. The codec in the modules folder is ONLY for Asterisk 13. Sorry if I didn’t make that clear. Asterisk 13 should handle just about anything you need.
@@InnovateAsterisk Okay...thank you for coming back that fast. I thought that this could be the reason. Quit happy it was not my fault. Great Tutorial, Mate!
It's worth mentioning that Asterisk versioning is not the same as most software that version 3 is newer and better than version 2 etc etc. Asterisk adopt like a branch style release system. So version 13 is as "up to date" as say 16 and 17, it's just that it's another branch, and is maintained by another team, this branch may even explore or solve different issues. I understand that big push for version 16 is to do multiple stream video conferences over WebRTC, something like that Google Meet and Zoom can do. So most of the older and more simple technology will be exactly as it was years ago.
Thanks for this video, it was very helpful. Just a note, I needed to disable optimizations to make the STRPTIME dialplan function work. It would only return 0 when Asterisk was compiled with optimizations enabled.
Thank you for this amazing tutorial! I followed the exact steps for Asterisk 16.13.0 and it works. I just have a question, does this git project that we cloned is okay for Asterisk 16.13.0? or we must clone what is compatible with it? From my side it works, but just wondering. Thank you in advance
Hi Conrad, thanks for the videos so educational, I am still having problems with Chan_SIP and hangingupp after 32 seconds of a call, ....is there any chance you can make a video configuring asterisk peers but with pjsip ? . Thanks again. Be safe.
There can be a number of reasons why a call may cut after 30 seconds, it can even be up to your network, and nothing to do with Asterisk. I’ll see what I can do with regards to a video topic on that. In the mean while take a look at: ruclips.net/video/mS28vfT8wJ8/видео.html (part 1 chan_sip) ruclips.net/video/azWUfSBz__s/видео.html (part 2 chan_pjsip)
This video shows how to install Asterisk on a Raspberry Pi. CentOS is different to Debian/Raspbian in many ways. Make sure you run contrib/scripts/install_prereq install first before you run ./configure --with-pjproject-bundled
Yes, its possible, but you have the change some settings. Firstly you cannot use Raspbian OS, so your next best option is Ubuntu. Next AWS will only offer you NATed connection inbound to your instance, so you have the let Asterisk know your network config. In sip.conf, add the lines (something like): localnet=172.16.0.0/12 ; Your AWS VPC externhost=ec2-aaa-bbb-ccc-ddd.eu-west-1.compute.amazonaws.com externrefresh=180 (Probably best to use a Elastic IP address, the following would work too) externaddr = aaa.bbb.ccc.ddd Also, have a good read: github.com/asterisk/asterisk/blob/1f78ee9d0f83bfeac2a73da99d526061a4437142/configs/samples/sip.conf.sample#L869
Great tutorial, just compiled Asterisk-18.3.
Few changes but this led me through the whole process nice and easy!
Thanks very much for this great series
Glad it helped!
Conrad - thanks for the shoutout and for picking up my Suggestion. I have tried several times to Setup Asterisk and I often failed. Your Videos took me up to the next Level. My very simple RPI-PBX is already running. I will now take an Espresso and watch this Video. All you Guys out there - Take Care and Stay Save!
Greetinz from Germany,
Gerd
All the best!
@@InnovateAsterisk Ok - here we go: I installed Asterisk 16, following your Tutorial and everything went fine. There is one Problem at least. Asterisk does not load the codec_opus_arm.so … It is available in the "/usr/lib64/modules/asterisk" -Folder but a "core Show Translation" does not list the Opus Codec. Do you have a Suggestion, what the Problem could be?
A "core show codecs" lists the Opus Code.
Yes, this the same as what I experienced. The codec in the modules folder is ONLY for Asterisk 13. Sorry if I didn’t make that clear. Asterisk 13 should handle just about anything you need.
@@InnovateAsterisk Okay...thank you for coming back that fast. I thought that this could be the reason. Quit happy it was not my fault. Great Tutorial, Mate!
It's worth mentioning that Asterisk versioning is not the same as most software that version 3 is newer and better than version 2 etc etc. Asterisk adopt like a branch style release system. So version 13 is as "up to date" as say 16 and 17, it's just that it's another branch, and is maintained by another team, this branch may even explore or solve different issues. I understand that big push for version 16 is to do multiple stream video conferences over WebRTC, something like that Google Meet and Zoom can do. So most of the older and more simple technology will be exactly as it was years ago.
Thanks for this video, it was very helpful. Just a note, I needed to disable optimizations to make the STRPTIME dialplan function work. It would only return 0 when Asterisk was compiled with optimizations enabled.
Thanks for the info!
Thank you for this amazing tutorial! I followed the exact steps for Asterisk 16.13.0 and it works.
I just have a question, does this git project that we cloned is okay for Asterisk 16.13.0? or we must clone what is compatible with it? From my side it works, but just wondering.
Thank you in advance
The same code worked for both Asterisk 13 and 16 +
Great...
Can you please make a new video using asterisk 20 on a intel based debian machine, please?
Great topic for another video. Just one note tho, Debian is very very similar to Ubuntu, so it would be very similar.
Hi Conrad, thanks for the videos so educational, I am still having problems with Chan_SIP and hangingupp after 32 seconds of a call, ....is there any chance you can make a video configuring asterisk peers but with pjsip ? . Thanks again. Be safe.
There can be a number of reasons why a call may cut after 30 seconds, it can even be up to your network, and nothing to do with Asterisk. I’ll see what I can do with regards to a video topic on that.
In the mean while take a look at:
ruclips.net/video/mS28vfT8wJ8/видео.html (part 1 chan_sip)
ruclips.net/video/azWUfSBz__s/видео.html (part 2 chan_pjsip)
I tried installing asterisk on centos6 and it’s showing me an “all error” message wen installing d pjproject wat do I do ... it’s frustrating
This video shows how to install Asterisk on a Raspberry Pi. CentOS is different to Debian/Raspbian in many ways. Make sure you run contrib/scripts/install_prereq install first before you run ./configure --with-pjproject-bundled
Does asterisk work in AWS?
I tried installing it but theres no audio or video being received
Yes, its possible, but you have the change some settings. Firstly you cannot use Raspbian OS, so your next best option is Ubuntu. Next AWS will only offer you NATed connection inbound to your instance, so you have the let Asterisk know your network config. In sip.conf, add the lines (something like):
localnet=172.16.0.0/12 ; Your AWS VPC
externhost=ec2-aaa-bbb-ccc-ddd.eu-west-1.compute.amazonaws.com
externrefresh=180
(Probably best to use a Elastic IP address, the following would work too)
externaddr = aaa.bbb.ccc.ddd
Also, have a good read:
github.com/asterisk/asterisk/blob/1f78ee9d0f83bfeac2a73da99d526061a4437142/configs/samples/sip.conf.sample#L869
0:36 in other word we are going to waste a lot of time.