Hi guys this looks like exactly what I need. How to I get the call.php file into freepbx and where is best to locate it? Any help is greatly appreciated! Thanks
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Freepbx Jolt Select Dial PBX - Chrome Plugin
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Incoming calls drop after 1 minute and 2 seconds
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Paging PBX 2 from PBX 1
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FreePBX to FreePBX SIP URI calling
Fixed it for you. 
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Reload failed because retrieve_conf encountered an error: 1
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Cisco 6851 3PCC (Third Party Call Control)
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Custom dialplan is not recording any calls
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Zulu login prompt when accessing UCP
Thanks for the tip on the services vs modules. I will definitely add to our internal documentation some references to using fwconsole as well.
I’m not so concerned about the possibility of a bug anymore, although I did a cursory search and didn’t find any existing bugs that seemed related.
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Custom dial plan for conference
Î want to use custom bridge and user profiles for one conference, so that the confbridge app in the dial plan looks like this:
exten => STARTMEETME,1,ConfBridge(5999,custombridge,customuser)
I could of course create the whole dial plan for this conference, however I would still like to be able to manage conference participants via conference pro module in UCP.
For this I assume I have to create the conference from the GUI, which also creates the dial plan for it.
What is the best way to overwrite freepbx generated dialplan for a confbridge?
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Registration failed and does not recover
I am occasionally getting the following… It pretty much takes a reboot to clear it.
Asterisk version 13.22.0 (from reports->asterisk info)
PBX firmware 12.7.6-1904-1.sng7 (from system admin)
Fully updated, both modules and system, not on edge track.
No idea how this maps Into current versions?
It starts In the middle of the night. no calls around the time. I notice in the morning. unregister/register doesn’t help. Stopping asterisk doesn’t help.
I can ping the host inbound1.
[2019-06-19 22:23:25] VERBOSE[6945] res_pjsip/pjsip_configuration.c: Endpoint vitel-inbound-pjsip is now Unreachable
[2019-06-19 22:23:25] VERBOSE[6945] res_pjsip/pjsip_options.c: Contact vitel-inbound-pjsip/sip:XXXXX@inbound1.vitelity.net is now Unreachable. RTT: 0.000 msec
[2019-06-19 22:24:31] WARNING[1534] res_pjsip_outbound_registration.c: No response received from 'sip:inbound1.vitelity.net' on registration attempt to 'sip:brkl_3355rack1@inbound1.vitelity.net', retrying in '60'
[2019-06-19 22:26:03] WARNING[6945] res_pjsip_outbound_registration.c: No response received from 'sip:inbound1.vitelity.net' on registration attempt to 'sip:brkl_3355rack1@inbound1.vitelity.net', retrying in '60'
[2019-06-19 22:27:35] WARNING[6945] res_pjsip_outbound_registration.c: No response received from 'sip:inbound1.vitelity.net' on registration attempt to 'sip:brkl_3355rack1@inbound1.vitelity.net', retrying in '60'
[2019-06-19 22:29:07] WARNING[1534] res_pjsip_outbound_registration.c: No response received from 'sip:inbound1.vitelity.net' on registration attempt to 'sip:brkl_3355rack1@inbound1.vitelity.net', retrying in '60'↧
FreePBX to FreePBX SIP URI calling
Two alternative approaches you might consider:
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Let someone else have the security headache. Get a trunk from e.g. Callcentric. Your account number is of the form 1777XXXXXXX. You register to them and anyone can call you with a SIP URI of
1777XXXXXXX@in.callcentric.com. If you use it for just that, there would be no cost, though you would probably fund the account for use as a backup trunk. On the client systems, define 00 as a Custom Extension with Dial string of
SIP/1777XXXXXXX@in.callcentric.com -
Use the PSTN. Set up a Flowroute trunk on each client system, as first priority for toll-free outgoing. Such calls are rated at zero and if your clients have significant traffic to toll-free it will give them some extra capacity (unrelated to using it for HelpDesk calls). Again, this could be no cost, though you would probably fund the account for general backup use. At your end, get a toll-free number. For this application, you could use a cheap one such as AlcazarNetworks ($2/mo. + $0.0125/min.) Don’t grumble about the 1 1/4 cents per minute; my guess is that the people on each end have a burdened cost 100 times that, so you’re talking ~0.5% overhead. And if the trunk fails, route advance to send the call the old way.
Neither of the above have any channel limitations at your traffic level.
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FreePBX to FreePBX SIP URI calling
I deliberately discounted that method because I’m not sure what the line length limitations are for the conf file. A sting of dozens of IP addresses may be a problem.
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System with Asterisk 13.26 waiting in yum logs, Can I stop it?
I have a few systems that had system upgrades set to yes. In the yum logs I can see they grabbed Asterisk 13.26. I know I can reboot, go to Asterisk 13.26 and then go back to 13.22 but is there any way to avoid the multiple switches and just load Asterisk 13.22 instead of 13.26 in the back end?
I tried to run yum upgrade again but the yum logs show no sign of grabbing 13.22 again.
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System with Asterisk 13.26 waiting in yum logs, Can I stop it?
Run asterisk-version-switch and choose either 13 or 16.
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System with Asterisk 13.26 waiting in yum logs, Can I stop it?
This will cause a slight disturbance yes?
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Create linux user in FreePBX's OS
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Twilio vs Les.Net
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System with Asterisk 13.26 waiting in yum logs, Can I stop it?
Asterisk is restarted in the process, so you need a maintenance window of a few minutes.
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System with Asterisk 13.26 waiting in yum logs, Can I stop it?
That’s what I thought. Thank you!
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Using ftp client is not possible
@Stewart1
Hi, I’ve been using scp for basic transfer needs, and I need Filezilla for collaboration environment. It seems that port 21 is blocked. I have it changed on a server machine, but it seems Freepbx won’t allow it, ast it seems getting back to the default blocked state.
When I yum install, I’ve got a base url error for sng-base/7/x86_64 too, so installing Winscp is not possible either too as I can’t use install commands. So in all scp and sftp are viable options if it gets to the point, but my Freepbx seems to need a handful of works before I actually work on customizing feature codes.
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