So our FreePBX v14 (asterisk v13) server died last week (only 10 months old HDD failure) and I had to rebuild it over the weekend. The backup was on the server and I had not gotten around to copying said backup to external flash drive or other location. So, unable to restore my backup. None-the-less, I have rebuilt the server with RAID and reinstalled/configured just about everything, except I am having a weird registration issue. My provider says the registrations are successful, however at the CLI, when I type ‘sip show registry’ it returns ‘0 registrations’. Users have informed me that incoming calls work, paging system works, however, outgoing calls do not-
freepbx*CLI> sip show registry
Host dnsmgr Username Refresh State Reg.Time
0 SIP registrations.
You should probably switch to the newer PJSIP stack. The old chan_sip module doesn’t handle DNS SRV/NAPTR records correctly. Maybe your problem goes away when you do that.
In FreePBX under Reports Asterisk Info it shows the Trunks are registered. When I access the Asterisk CLI and type ‘sip show registry’ it returns ‘0 registrations’
I restarted the asterisk core and still nothing.
Because that’s only for chan_sip. The equivalent for pjsip is pjsip show registrations
Though I know nothing about NexVortex, very few providers require registration to make outbound calls, so it’s unlikely that even if you are failing to register it’s related to your issue.
At the Asterisk command prompt, type pjsip set logger on
make a failing attempt at an outgoing call, paste the Asterisk log (not the console output) for the call at pastebin.freepbx.org and post the link here.
We made several call attempts today and NexVortex did not see any of them hit their system. Never got outside the PBX. Yes there is an Outbound Route(s) - fully configured.
One for emergency, one blocked for long distance prefixes off shore and the final is our Outbound route for all other calls.
Possibly the phones got incorrectly provisioned, or are not reaching Asterisk for some other reason. What does the caller hear? What displays on the device? Does anything get added to /var/log/asterisk/full when you attempt a call?
hey guys,
Everything is PJSIP (Extensions and Trunks) Trunks are registered.
I re-provisioned all the phones this morning. Save-rebuild configs-update phones.
Endpoints still unable to make outgoing calls.
When they dial my number they hear the DTMF echo and then the extension just sits there. It shows, my name and my phone number. On the display too the left of the number they dialed there is an icon of a phone with a red x or slash through it. After 10 seconds it drops the call attempt. I’ll see if I can get anything from the log and post it. But prior attempts to grep the log came up empty. The AOR shows all the extensions so I know they are connected.
You asked for a debug of the outbound routes - how do I do that? Sorry I am not very experienced.
Thank you,
Chris
Trunks registered but still no calls.
I erased and re-configured 1 extension to test and still nothing.
Disabled firewall and tried still nothing.
Logs empty
Please help.
Hey Stewart,
I tried the following-
disabled Sangoma Firewall. Checked intrusion detection - nothing noted.
Nothing is added to the logs.
I checked and updated my Extensions, Extensions Mapping and Global Endpoint Settings.
I am at a loss. Now there is no delay when user dials a number, it drops almost immediately with out any delay. Nothing is noted in the logs when user calls.
using the following: grep full
Any ideas or things to check?
AOR’s look good.
Thank you,
Chris
here is the verbose call log for a dropped call (at the bottom) they just tried calling my cell phone. Call dropped immediately. Something about it being restricted. Not sure why. Please let me know what you think.
Indeed, the log shows that extension 306 is restricted from calling the 1925 number. I assume that you are using Extension Routing and not using Class of Service. If that’s not the case, provide details.
On the Additional Settings tab for the Outbound Route in question, confirm that 306 is in the list of Allowed extensions. After Submit and Apply Config, restart Asterisk. If you still have trouble, post screenshots of your Outbound Routes, including the Dial Patterns and Additional Settings tabs.
Hi Stewart,
Thank you. I just deleted the outbound routes and re-created them. I corrected a couple issues now they calls are not dropped but they get Asterisk Allison saying the call could not be completed as dialed, so some progress. Here is a new pastebin.
Yes all my extensions are in the allowed list. i’ll see about some screen shots of the config.