I am glad that I found this video. Following your guide (in Dec of 2018) I found everything still very relevant and was able to get my trial up and running in no time. Thanks for the time you put into making it.
Do you have any tutorial getting connected Twilio and issabel PBX? I have looked for many pages and I could not have found anything. Which parameters do I have to use in the outgoing field?
I just followed along with this video and did exactly what you did, but I'm not able to make an inbound call to my Twilio number. Calling outbound works great, but I get a fast busy signal when I try to call in. The Twilio call logs show my attempted inbound call with a status of "Failed" and the Asterisk logs give me this error: [2017-06-30 18:34:09] NOTICE[5686]: res_pjsip/pjsip_distributor.c:526 log_failed_request: Request 'INVITE' from '' failed for '54.172.60.3:5060' (callid: 0b319be48ac83b0d66389524bd65284a@0.0.0.0) - No matching endpoint found Any idea what I'm doing wrong? I have the inbound call set to ring my extension and as far as I can tell, I have my extension set up correctly. It dials out anyway! ETA: Nevermind - I finally got it working! It was an issue with the ports. Twilio was trying to connect using port 5060, but the current default installation of FreePBX has chan_sip using 5160 and chan_pjsip using 5060. Twilio doesn't seem to work well with chan_pjsip, so I had to find the settings to swap the ports between sip and pjsip.
Alisa, Glad you got it figured out! Thanks for updating the comments with your solution. The install I was running was Freepbx13 asterisk 11 using chan_sip. For others who get caught with the same issue... To force chan_sip (if you installed asterisk 13) go to Settings > Advanced Settings > then change "Sip Channel Driver" to chan_sip. Next go to Settings > Asterisk Sip Settings and update the Chan_Sip Bind Port to 5060 and the TLS Bind Port to 5061. You will also need to update the chan_pjsip Ports to 5160 and 5161. If you installed asterisk 11 from the start then the chan_pjsip tab will not appear in the Asterisk Sip Settings menu, but you may still have to update the ports in the chan_sip tab.
It is outdated. I got it working and it wasn't working smoothly... I haven't got it working yet but they have pdf documentation on there website for FreePBX 15 version.... Totally different ball game from the tutorial. They say you need to use pjsip now.
I hear nothing when call my number and i get this error: Error communicating with your SIP communications infrastructure: Request timeout with sip : ..... And I get 401 unauthenticated on Invite. credentials are correct I tried everything..
![[part 10] Setting up SIP trunk on your FreePBX system so it can talk to the phone company](http://i.ytimg.com/vi/T4qj4vq2B-0/mqdefault.jpg)
I am glad that I found this video. Following your guide (in Dec of 2018) I found everything still very relevant and was able to get my trial up and running in no time. Thanks for the time you put into making it.
.
You are the man! Helped me get it setup in minutes!!!
Thanks for this. Helped me alot
Thanks got my Twilio trunk working with the help of your video. Great job keep going
No problem 👍
Really useful video. Thank you!
Do you have any tutorial getting connected Twilio and issabel PBX? I have looked for many pages and I could not have found anything. Which parameters do I have to use in the outgoing field?
origination sip:ip address
and
termination access ip address will be same or different?
I just followed along with this video and did exactly what you did, but I'm not able to make an inbound call to my Twilio number. Calling outbound works great, but I get a fast busy signal when I try to call in. The Twilio call logs show my attempted inbound call with a status of "Failed" and the Asterisk logs give me this error:
[2017-06-30 18:34:09] NOTICE[5686]: res_pjsip/pjsip_distributor.c:526 log_failed_request: Request 'INVITE' from '' failed for '54.172.60.3:5060' (callid: 0b319be48ac83b0d66389524bd65284a@0.0.0.0) - No matching endpoint found
Any idea what I'm doing wrong? I have the inbound call set to ring my extension and as far as I can tell, I have my extension set up correctly. It dials out anyway!
ETA: Nevermind - I finally got it working! It was an issue with the ports. Twilio was trying to connect using port 5060, but the current default installation of FreePBX has chan_sip using 5160 and chan_pjsip using 5060. Twilio doesn't seem to work well with chan_pjsip, so I had to find the settings to swap the ports between sip and pjsip.
Alisa,
Glad you got it figured out! Thanks for updating the comments with your solution.
The install I was running was Freepbx13 asterisk 11 using chan_sip.
For others who get caught with the same issue...
To force chan_sip (if you installed asterisk 13) go to Settings > Advanced Settings > then change "Sip Channel Driver" to chan_sip.
Next go to Settings > Asterisk Sip Settings and update the Chan_Sip Bind Port to 5060 and the TLS Bind Port to 5061.
You will also need to update the chan_pjsip Ports to 5160 and 5161.
If you installed asterisk 11 from the start then the chan_pjsip tab will not appear in the Asterisk Sip Settings menu, but you may still have to update the ports in the chan_sip tab.
I can only get inbound calls to work if I enable allow anonymous sip calls. What am I missing that it doesn't work when disabling allow anonymous?
Can u share your steps for install all in ubuntu 16?
does this still work and does it work on kali linux
What is the user for the AMI. When I try to SSH into it it asks for a password
THis must be outdated as my Twilio account looks nothing like this and this video DID NOT HELP AT ALL
It is outdated. I got it working and it wasn't working smoothly... I haven't got it working yet but they have pdf documentation on there website for FreePBX 15 version.... Totally different ball game from the tutorial. They say you need to use pjsip now.
I hear nothing when call my number and i get this error: Error communicating with your SIP communications infrastructure: Request timeout with sip : .....
And I get 401 unauthenticated on Invite. credentials are correct I tried everything..