May 19, 2015, 11:04am
I’ve just install FreePBX from distro FreePBX-64bit-10.13.66.img
And now I have problems with adding new users.
I’ve registered SIP extension 723 and created same user connected to that extension.
Changes aplied without errors
SIP client can’t connect with error:
SIP/2.0 403 Forbidden (policy)
SIP/2.0 404 Not found
Server side error:
[2015-05-19 11:40:10] NOTICE: chan_sip.c:28235 handle_request_register: Registration from '<sip:
[email protected]>' failed for '10.10.1.123:5060' - No matching peer found
At the server ssh session
rasterisk show to me:
localhost*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
What can I do to fix this problem?
May 19, 2015, 12:05pm
You need to use the right ports. You’ve added chan sip extensions but are trying to register them on pjsip ports. The ports for Cham sip are 5061 by default
May 19, 2015, 12:16pm
The Bind Port for Chan sip in my case is 5060
But I can’t enter the Chan pjsip setting - page error.
Invalid argument supplied for foreach()
3. Whoops\Run handleError
1. Sipsettings myShowPage
May 19, 2015, 12:40pm
The problem is that no new users appears in
After adding they manually to the file, phones connected normally, but users doesn’t appear in GUI
May 19, 2015, 12:50pm
Can I upgrade to Asterisk 12 or 13 from the version 11.17.1?
And is it there new versions of FreePBX (the mine is 10.13.66-2)?
May upgrade 'll fix problems…
May 19, 2015, 12:51pm
You are using alpha software. Best to use stable. 6.12.x releases.
May 19, 2015, 12:56pm
Is it possible to downgrade to 6.12 without full reinstall of system?
Or troubles with difference in config files and DB doesn’t possible to fix in automated way?
May 19, 2015, 1:40pm
what type of device are you adding pjsip or chansip
May 19, 2015, 3:25pm
Few types of Grandstream phones.
I think I’ll try to install 6.12.x version and enter all the data manually.
Now I found only one solution: using *_custom configs and editing them manually without PBX :((