Hi, I’m currently in the middle of transitioning my VOIP provider from Sipgate to Xinix.
The company itself has provided me with a Grandstream HT701 ATA and I already have a HT812, both of these are able to register just fine.
However on FreePBX I’m unable to register, with errors like: No response received from ‘sip:(username)@(domain)’ on registration attempt to ‘sip:(username)@(Domain):5060’, retrying in ‘60’
The Details I have been provided are: Domain/Proxy: (domain) Outbound Proxy: sip(dot)sipdesk(dot)net Username: (username) Password: (password)
In the PJSIP settings, I have tried different settings including: username/authusername: (username) domain: (domain) server port: 5060 (taken from grandstream) outbound proxy: sip:sip(dot)sipdesk(dot)net:5060;lr
The outbound proxy was taken from the syslog dump of the Grandstream.
no luck unfortunately, still no response: No response received from ‘sip:(domain)’ on registration attempt to ‘sip:(username)@(domain)’, retrying in ‘60’
I’m puzzled. Just for laughs, I set up a fake Xinix trunk with Username: 1234, Secret: 4321, SIP Server: example.com
and got (as expected): [2023-11-07 06:34:40] WARNING[12260] res_pjsip_outbound_registration.c: 403 Forbidden fatal response received from 'sip:example.com' on registration attempt to 'sip:[email protected]', retrying in '30' seconds
So, possibly a network or DNS issue; confirm that you can ping sip.sipdesk.net from the PBX.
If that’s ok, recheck that Outbound Proxy is correctly set (note two occurences of backslash semicolon).
If no luck, at the Asterisk command prompt type pjsip set logger on
wait for a failed registration attempt, and post the REGISTER request (including the line showing where it was sent).
That’s an OPTIONS request, not a REGISTER request, but if you have confirmed that there is also no response to OPTIONS, it’s just as good. If there was a response, please post it, as well as a REGISTER.
Confirm that (my IP) means your public IPv4 address, same as you see at whatismyip.com .
Next, I suspect either a SIP ALG problem (make sure that you disabled anything related to SIP in your firewall router), or the source port number is being rewritten and sipdesk is ignoring the rport request.
Next, describe your networking setup. Router/firewall make/model? Any VoIP-related settings? Does it have your public IPv4 address on its WAN interface? If not, please explain (connected to ISP-supplied gateway, ISP does CGNAT, etc.)
Make sure that you use bridged networking (PBX IP address is in same subnet as host). If you still have trouble, post details about virtualization setup.
So far, the closest I’ve got with the main distro is using Asterisk 19 with a clean install and starting from scratch, but even this is very hit or miss and when it is working I can make outgoing calls but incoming give me an error on “invite” saying “failed to authenticate”.
Next thing to try might be DMZing the VM, at least temporarily, to make sure it’s not a firewall issue.
I’m also contemplating trying a Rube Goldberg option, using an old ATA with an FXO port as a bridge for another Grandstream ATA (the former has a broken FXS port).
My setup is a proxmox host with a quad Intel NIC PCI-E card. For the VMs each NIC uses PCI-E passthrough. One VM is my firewall pfSense which has 2 ports, one for WAN and LAN. Another port is used for Managing Proxmox and finally one is dedicated to FreePBX.