Hi all, first-time poster here and also first time FreePBX user.
I’ve just set up FreePBX 15.0.16.78 and connected a Cisco SPA504g which get a dial tone.
I’ve set-up a Trunk which points towards Sipgate UK and is online, I’m having issues making and receiving calls which may be down to the dial plan, I will try and list out my configuration and trouble shouting below with the hope that it will help you help me resolve this issue.
I can see logs from outgoing calls but no logs incoming
Her goes
Trunk Configuration
Under General, I just have the trunk name Sipgate
Dial Number Manipulation
I have ‘X.’ without the ‘’
Currently it is best to use PJSIP for SIP rather than the legacy chan_sip driver, especially for brand new installs.
I would recommend disabling chan_sip (Settings - Advanced Settings - set SIP driver to chan_pjsip only). Do this after you have removed your extension and trunk configurations that currently use chan_sip. Then create the configs again with PJSIP.
This may sound like a lot of work but it should be beneficial to you in the end, and this is the best time to do it–before you have set up a lot of trunks and extensions.
Nothing in your screen grab looks abnormal, except the 503 which it likely because sipgate rejected the call. I’m guessing that your sipgate trunk is using chan_sip so those transactions didn’t appear in pjsip history.
Please post a complete debug:
At the Asterisk command prompt, type pjsip set logger on sip set debug on
and make a failing test call. The regular Asterisk log will now include both pjsip and chan_sip traces, in addition to the normal entries. You can access it at Reports -> Asterisk Logfiles, or directly at /var/log/asterisk/full
Take the entire log for the call, redact as desired, paste it at pastebin.freepbx.org and post the link here.
Outbound calls to 9123 failed because your Outbound Routes do not match that pattern.
Calls to 123 failed because the trunk settings need to include a From User setting with your sipgate username.
Inbound calls are failing because in Asterisk SIP Settings, External Address and Local Networks are not properly set. Restart Asterisk after changing these.
After fixing the above, if you still have trouble, post new logs.
Thanks, for your responses, I’ve quite a bit to learn with regards to this system, I’m going to carry out a clean install and move away from a VM and then build from the ground up again, hopefully, my next post will be to say it’s been successful.
