Can you do an introduction into the basics of PBX systems, like what is needed to make outbound and inbound calls, what are trunks etc? The basics of the PBX infrastructures, the need-to-knows and stuff. Great content by the way!
@@AwesomeOpenSource thank you so much. Just want to learn the basics/fundamentals of how to get started. I think I have an idea but its not as clear as I need it to be. I've administered asterisk, elastix and fusionpbx systems but I need to understand it more.
It's been a really long time since I made this, so I don't really remember any more, but I'd suggest going over to the FusioPBX forums and asking there.
Thanks! I really enjoy your videos and all of these open source projects you dig up! I have an issue, I for the life of me cant get an outgoing call to function, gxp2170 or with various softphone apps. Just states no response. Any tips for a sip nooooob?
I would check out the VOIP Info forums www.voip-info.org/forum/. Lots of great information and users there that may can help. This video is getting fairly old, so the way you setup outgoing may have changed some. I know at one point I noticed it added some routes (that's what I call them) in the outgoing part. You can see it in the setup of the outgoing rules. Basically I had to remove it, as it said, just send it to nowhere. It was odd for sure.
when you receive a phone call, do you get the caller id of the calling person, or is it just the end point number calling you in your system? i have a gateway, that is forwarding the call to my sip number, the from field has the caller id of the client, but the id i receive is 2001. Any idea, is it the same issue on your side?
You can set it up to provide the caller ID of the calling person or the internal call. You can also set your caller ID to be your direct line, or the number of the main line (so to speak).
I started to mess with fusionpbx just to see if I can get it to work. I'll have to check out the rest of your videos to see if maybe you have an answer to my issue. I have the PBX at a remote location and I am trying to setup an IP phone to go to that location. I am able to provision the phones that I have here through the PBX but when I tried to dial extension to extension it will call the pbx but the other extension never rings so trying to figure out what setting or software I need to add into this to make the magic happen.
@@AwesomeOpenSource no worries - I found the answer to my issue. Had to install a package on our firewall and with that it started working. Now I just have to figure out how to change the provisioning template for the phones and i'll be golden. Thank you for replying.
That would be a question for the FusionPBX Documentation, or the Fusion PBX team. if you mean only allowing a specified IP to connect to the system, I wouldn't call that authentication specifically, but it can be accomplished with firewall rules in IP tables. IP Spoofing, would be a fairly easy workaround through, so not overly secure on it's own, but could be part of a more layered security model. would probably be better to run it on a VPN, and then have users / devices join that VPN to connect.
after every step what do we have to do in twinkle because i try to make outbund phone call to my personal phone and i get this error Line 1: call failed. 480 Temporarily Unavailable
Assuming you can call extension to extension, then something in the outbound route maybe messed up. You can see the logs in the web UI. Worth looking there.
I don't anymore. I set it up as a simple way for my business to route calls to me when I worked for my last company. I no longer need this with my current company, so I've no need for the system at this time.
Let me see what I can do. I did setup LetsEncrypt for my production setyp. It does fail to auto renew occasionally, and I have to go manually do it still, but it generally does what it's supposed to.
@@AwesomeOpenSource yeah, same here, i just re run it as well. let me know if you find a solution or post part 3 of this series showing ssl part. anyways suscribed. cheers
@@AwesomeOpenSource Not sure what distro you run but for auto renew Edit file: nano /etc/cron.d/certbot 0 */12 * * * root test -x /usr/bin/certbot -a \! -d /run/systemd/system && perl -e 'sleep int(rand(43200))' && certbot -q renew --renew-hook "systemctl reload apache2"
This is exactly what i need! Thank you so much!
You are so welcome!
Can you do an introduction into the basics of PBX systems, like what is needed to make outbound and inbound calls, what are trunks etc? The basics of the PBX infrastructures, the need-to-knows and stuff. Great content by the way!
Let me research it more, as I'm not a PBX expert by any means.
@@AwesomeOpenSource thank you so much. Just want to learn the basics/fundamentals of how to get started. I think I have an idea but its not as clear as I need it to be. I've administered asterisk, elastix and fusionpbx systems but I need to understand it more.
When creating the Gateway at minute 10:25, how can you change or configure the Contact field so that a very long string of characters does not appear.
It's been a really long time since I made this, so I don't really remember any more, but I'd suggest going over to the FusioPBX forums and asking there.
Will you do another video showing how to setup TLS and SRTP on FusionPBX?
Let me see what I can do. It may be a while.
Good video
Thank you so much.
Thanks! I really enjoy your videos and all of these open source projects you dig up! I have an issue, I for the life of me cant get an outgoing call to function, gxp2170 or with various softphone apps. Just states no response. Any tips for a sip nooooob?
I would check out the VOIP Info forums www.voip-info.org/forum/. Lots of great information and users there that may can help. This video is getting fairly old, so the way you setup outgoing may have changed some. I know at one point I noticed it added some routes (that's what I call them) in the outgoing part. You can see it in the setup of the outgoing rules. Basically I had to remove it, as it said, just send it to nowhere. It was odd for sure.
when you receive a phone call, do you get the caller id of the calling person, or is it just the end point number calling you in your system? i have a gateway, that is forwarding the call to my sip number, the from field has the caller id of the client, but the id i receive is 2001. Any idea, is it the same issue on your side?
You can set it up to provide the caller ID of the calling person or the internal call. You can also set your caller ID to be your direct line, or the number of the main line (so to speak).
I started to mess with fusionpbx just to see if I can get it to work. I'll have to check out the rest of your videos to see if maybe you have an answer to my issue. I have the PBX at a remote location and I am trying to setup an IP phone to go to that location. I am able to provision the phones that I have here through the PBX but when I tried to dial extension to extension it will call the pbx but the other extension never rings so trying to figure out what setting or software I need to add into this to make the magic happen.
Hmmmm, strange for sre. Keep in mind this video is getting a bit old, so things may have changed in FusionPBX since it was made.
@@AwesomeOpenSource no worries - I found the answer to my issue. Had to install a package on our firewall and with that it started working. Now I just have to figure out how to change the provisioning template for the phones and i'll be golden. Thank you for replying.
Imagine taking 4G usb modem and turning it into sms and phone call gateway, instead of having to rely on third party voip numbers...
How can i create gateway using IP Authentication
That would be a question for the FusionPBX Documentation, or the Fusion PBX team. if you mean only allowing a specified IP to connect to the system, I wouldn't call that authentication specifically, but it can be accomplished with firewall rules in IP tables. IP Spoofing, would be a fairly easy workaround through, so not overly secure on it's own, but could be part of a more layered security model. would probably be better to run it on a VPN, and then have users / devices join that VPN to connect.
for some reason i cant make incoming calls i can call out just not in
That is odd, but it could be something odd in the dial plan.
@@AwesomeOpenSource yea I was going to try to tackle it more tomorrow
after every step what do we have to do in twinkle because i try to make outbund phone call to my personal phone and i get this error Line 1: call failed.
480 Temporarily Unavailable
Assuming you can call extension to extension, then something in the outbound route maybe messed up. You can see the logs in the web UI. Worth looking there.
@@AwesomeOpenSource sorry mean fusionPBX, i did set up the outbound how u did with voicetel configuration
@@sheilasquires5065 There could be some ports that need to be opened as well, but does it show ViceTel as reged when you set it up on the Gateway?
Hello, you still using fusionpbx?
I don't anymore. I set it up as a simple way for my business to route calls to me when I worked for my last company. I no longer need this with my current company, so I've no need for the system at this time.
@@AwesomeOpenSource thank you to reply!
Isn't the default sip port, 5060? not 5080?
Mmmmm... I’ll have to go back and check. You may be rightz
@@AwesomeOpenSource Definitly right 5060 is for registration but RTP(for audio) ports usually go from 9000 - 11000
HI, can you show how to set lets encrypt auto renew?
Let me see what I can do. I did setup LetsEncrypt for my production setyp. It does fail to auto renew occasionally, and I have to go manually do it still, but it generally does what it's supposed to.
@@AwesomeOpenSource yeah, same here, i just re run it as well. let me know if you find a solution or post part 3 of this series showing ssl part. anyways suscribed. cheers
@@AwesomeOpenSource Not sure what distro you run but for auto renew
Edit file:
nano /etc/cron.d/certbot
0 */12 * * * root test -x /usr/bin/certbot -a \! -d /run/systemd/system && perl -e 'sleep int(rand(43200))' && certbot -q renew --renew-hook "systemctl reload apache2"
to run a test, just hit this comand
certbot renew --dry-run
@@armandwolfaardt7721 there is not cerbot file to edit...?