Trouble configuring SIP Trunk for Portuguese Altice (Enterprise) GPON


As a very new user to FreePBX, but through tutorials, lots of reading and some trial and error, i managed to configure extensions, make internal calls and even configure an external VOIP provider and make external calls using that.

However, when configuring a SIP Trunk to our local provider, i seem to have hit a wall…
Evertthing seems to be well configured, but i get the following message when i restart the services (fwconsole restart) and enter verbose mode (asterisk -rvvv):
Contact Altice/sip:XXX.XXX.XXX.XXX:5060 is now Unreachable. RTT: 0.000 msec

We have an Alcatel OXE which we are trying to replace, and it does work, so network problems don’t seem to be the case.

I’m basically out of ideas.
Maybe someone has a fresh perspective?


After further digging, i’ve found the following responses:
[2020-11-18 11:06:31] WARNING[1901]: res_pjsip_outbound_registration.c:829 schedule_retry: No response received from ‘sip:XXX.XXX.XXX.XXX:5060’ on registration attempt to ‘sip:[email protected]:5060’, retrying in ‘60’

And if i try to place a call, i get a busy message, and also this output:
[2020-11-18 11:10:46] ERROR[1901]: res_pjsip.c:3562 ast_sip_create_dialog_uac: Endpoint ‘Altice’: Could not create dialog to invalid URI ‘Altice’. Is endpoint registered and reachable?
[2020-11-18 11:10:46] ERROR[1901]: chan_pjsip.c:2687 request: Failed to create outgoing session to endpoint ‘Altice’
[2020-11-18 11:10:46] WARNING[3695][C-00000001]: app_dial.c:2576 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)

The server is definitely reachable.
So, the prblem is either:
-Bad config (i used PJSIP, but i should try with SIP?..)
-Network problem (incoming)

If you are using IP Authentication, set your Authentication to “None”.

Hi Dave!

Thanks for your reply.

That was actually my first try.
I only changed it to send blank uername and password as a test.


If you are using username/password, you need to make sure those are correct.

Your ITSP can’t send traffic to your PBX, so make sure your firewall is tranferring the right ports to the PBX.

To me, this looks like a networking/NAT problem. Make sure you have all of your configuration information for NAT and the external connections correct.

Our service does not require user/pass.
The ITSP has a Cisco Router in our datacenter that creates a VPN directly into their datacenter.
We do not have the credentials of the router.
There a static route in our core switch that routes all traffic destined to the ITSP server via the router.
I managed to capture packets going to the Cisco Router and they are being sent by FreePBX. But nothing comes back as a response.
My 2 best guesses are:
-Since we already have one working Alcatel System, maybe the new connections are being rejected… (i’m going to try and shut it down later and test again, also maybe set the same ip for freepbx, because there might be a route in the cisco router forcing everything into the Alcatel System, but i don’t think it’s the case because traffic manages to flow into the Alcatel phones directly after a call is placed…)
-The ITSP is filtering every other MAC Address other than the Alcatel’s…

If shuting down the Alcatel system does not solve the issue, i’m going to open the case with the ITSP…

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