Able to make outgoing calls, but unable to receive calls


I am using Asterisk 16 under a FreePBX GUI. One of my SIP providers gives a separate “authentication name” and “username”, and after some troubleshooting and help from others I’ve managed to register the SIP with the Asterisk trunk and I can make outgoing calls. However, incoming calls fail. It rings about 5 times but then it just gets redirected to my SIP provider’s own voicemail box. My PBX connected phones do not ring.

Registration string I’m using that works: [authname]:[password]@[host domain]/[username]

PEER Details that work (UUUU=username, AAAA=auth name, PPPP=password, DDDD=SIP provider’s hostname):


It appears that I may need to tweak some of these settings, or fill up the USER Details section and USER Context.

Please see the following Asterisk error log for what happens when I try an inbound call (PBX server hosted on AWS):

Thank you!

Please note we can only really answer this in Asterisk terms, not FreePBX ones.

Neither of your logs contains any call attempts.

Please confirm that [PBX IP] is a public IP address.

That is what shows up on the logs when I try to make an incoming call. It just goes straight to the SIP provider’s own voice mail.

[PBX IP] is a static public IP address that sits on AWS.

Edit: to add some context, I have two different SIP service providers. One of the providers is fully set up on my PBX and everything works perfectly. This is the issue with the second SIP provider.

The provider isn’t sending the calls to you. If you have successfully registered, and the contact address is correct, you may need more help from the provider, although it is possible that you have firewall problems. The firewall may be creating a temporary opening for replies, but might not be open for unsolicited requests.

Thanks for the quick response.

It shows successfully registered. Unfortunately the provider’s support is not forthcoming on this issue.

I’ve examined all my firewalls (AWS firewall, fail2ban, FreePBX’s firewall), all have exceptions for the SIP provider’s IP address for 5060 and 10000-30000. I’ve configured chan sip for 5060 as well.

Also my current setup actually works perfectly with a second provider sitting on the same machine. Additionally, if I use softphones or deskphones to connect directly to the first provider (the problematic one), I am able to make outgoing and incoming calls perfectly fine as well. For the PBX now only outgoing calls work for the first provider.

‘tcpdump’ may yield some clues. If tcpdump shows you are receiving the INVITE, then enabling SIP debugging on the Asterisk console and bumping up debug and verbose settings > 2 may help.

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