I have been running a Trixbox PBX since 2009. Hard drive finally went out and was time to upgrade hardware and software. Have loaded Ubuntu 16.04 LTS and Asterisk 13.17.0. I am logged into the web interface gui from another computer and have set up the basics. I have two extensions 6013 and 6099 configured and showing up in “sip show peers”. On the web interface on the system status tab, both extensions show as active, and my trunk is showing as registered. Now the problem. I can not make, or receive any calls. I get “the person you are calling is unavailable” message. it happens while trying to call external, as well as calling from extension to extension internal. I don’t know where to look to figure this out. If someone could please help it would be greatly appreciated. My business phones are down, and I am losing money. I can provide any other info you may need. Thanks in advance. Sam
In the CLI I am getting
Notice[C-00000072]: chan_sip.c:26401 handle_request_invite: Call from ‘myaccountnumber’ (myprovideripaddress:5060) to extension’+174xxxxxxxx’ rejected because extension not found in context ‘DID_myaccountnumber’,
NOTICE: chan_sip.c:15875 sip_reg_timeout: – Registration for ‘6099@dynamic’ timed out, trying again (attempt #7)
Are you migrating to something using FreePBX as a GUI on top of Asterisk? Asterisk itself doesn’t provide a GUI. If so, you might want to check in with https://community.freepbx.org for assistance with FreePBX.
This is a fresh install of Ubuntu 16.04 Server LTS on a brand new computer with 2.4ghz 7th Gen i3 processor, 8 g ram and 1 TB hard drive. I then compiled Asterisk 13.17.0 to run in CLI environment. I installed
Asterisk GUI-version : SVN-branch-2.0-r5220. I am using a web interface along with CLI until I get this figured out. Sometimes I miss something in CLI, but then catch it in the gui, and visca versa.
17:23:00 up 1:56, 2 users,
Load Average: 0.95, 0.89, 0.81
Asterisk GUI-version : SVN-branch-2.0-r5220
Server Date & TimeZone:Wed Jul 19 17:23:00 EDT 2017
re: Asterisk-GUI, see:
Thats fine, its only temporary while I figure out the dial plan.
With my previous version (ie…OLD) calls would come trough th trunk to a section caled inbound routes. This is where all my DID numbers were located. from there they would go to any announcements, time conditions, queues, ring groups, etc…
The Asterisk system is just a different beast and I am trying to figure out the sequence of events to make it work.
What are others using as an FOP Operator Panel?
Here is my one extension 6099 call plan. Is there anything wrong? Have tried host=dynamic and host=static with same results.
Since you were using asterisk-gui, that also means you ended up using users.conf in lieu of pjsip.conf (or sip.conf) and voicemail.conf, etc. to try to configure endpoints. users.conf is also a dead thing.