Asterisk Set Up Trunk
I have been struggling now for days with setting up Asterisk and Twilio to work with Elastic SIP Trunk. When I dial the connected Twilio number i get a busy tone. It seem to be correctly configured in the SIP Elastic Trunk on Twilio, so the issue is probably somewhere in my config files. I'm not very used to Asterisk. From the debug feature in Asterisk I can read the following relevant information:
If we are using username/password authentication with our provider then we need to set our trunk up manually in Asterisk. We will give that trunk a name and then use that name, instead of the IP address, in the A2Billing trunk setup screen above. Here’s an example of setting up a trunk with switch2voip.us. Many providers will be similar and provide the connection details you need to enter. These settings go in the file /etc/asterisk/sip.conf. @u2communications said in Setting up a SIP trunk in FreePBX 13: If you find yourself still trying to get incoming calls working after several hours (like me), be advised that the default DID settings on. 2019-6-16 How to set up CallCentric as a trunk on FreePBX and Asterisk. Pingback: » CallCentric trunk setup with Asterisk/FreePBX « Gregory Reese Research elmohem 20 March 2013 at 6:55 am. I do not hear any message I must hear enter-pin-number.
And
My relevant configuration is
sip.conf
extensions.conf
I guess it has something to do with the Unauthorized error, however I have set up Twilio to accept my server IP and tho not authorize with username and password. To mention is that it works for my outbound routing. The issue is probably not related to internal routing as it works to call between my extensions. Also, all of my clients are behind NAT but the server is not behind NAT.
Asterisk Set Up Trunk Template
What is wrong with my config? Does anyone have a working example with twilio and asterisk?
1 Answer
Elephant Statue With Trunk Up
Forget 'SIP Trunking' configuration. These calls you´re receiving from 'SIPOut API servers', that´s the reason for the 'No matching peer'. sip.conf needs another type of peer to handle with SIPOut calls. Google it.
