I am very new to FreePBX but have used it at a local level for deploying extensions. I have been recently trying to use it more professionally so am trying to link it with our SIP Trunk. We have a trunk with yay dot com who have been brilliant so far.
I am trying to setup the trunks so I can make outgoing and receive incoming calls. I’ve followed their guide as shown online and contacted their support.
Their support said that my server is responding with a 407 Proxy Authentication Required to our incoming calls (INVITEs). They said they don’t support authentication on trunk calls and that I will need to configure freepbx to accept calls without authentication. Unfortunately I am very new to FreePBX and VOIP in general so this may seem like a bit of a dumb question!
Thanks in advanced for any support it is greatly appreciated!
Using chansip or pjsip trunks?
Are you using https://www.yay.com/faq/cloud-pbx/freepbx-config/ ? If not, please post screenshots of relevant settings (mask personal info).
Are outbound calls ok?
On an attempted inbound call, what (if anything) appears in Asterisk log?
I’m using chan_sip at the moment. I did try with pjsip yesterday but again didn’t have much luck.
On chansip, the parameter
insecure=invite in your peer details will prevent your server from challenging incoming Invites for authentication.
I think, but I am not 100% sure, that
remotesecret does the same and is the more modern version of that.
remotesecret=password instead of secret and see if you can receive calls.
If not, then go back to secret and use
Yes I was following that exact guide but couldn’t! post as I’m a new user. I have attached a few screenshots I hope they help. No I can’t make outbound calls either, I just get a recording saying your call cannot be completed as dialled please check the number and dial again. I cant see anything in the log but have also attached the top half.
This is not the log of a call.
sip set debug on and then place a call.
sorry, I’m very new to this. I’m using the web gui where abouts would I find this option?
Not from the GUI.
You go to the Asterisk Command Line.
SSH into your server and enter command
The enter command
sip set debug on.
Then place a call and you will see the full flow of SIP messages.
Your inbound calls are going to your pjsip port and not chansip.
Have you deleted the pjsip trunk?
oh, yes I have deleted the pjsip trunk.
Try the register string which you have left blank in your trunk settings.
See if you get the trunk to register.
The config example is in the link stewart1 posted.
disallow= and allow=
need to be on separate lines.
You might have deleted the pjsip trunk, but you will need a chan-sip trunk responsive to and from yay dot com on the port that they are set to use. Currently it looks like they are sending calls to port 5060 , which in your deployment is going to be answered by pjsip and as you see , pjsip doesn’t know what to do with them.
Perhaps adding port=5160 (or whatever) to your chan-sip trunk’s inbound and outbound settings?
In case you don’t know… By default PJSIP is on 5060 ChanSIP on 5160, you can change the ports, but it requires a Asterisk restart.