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For reference, the TILTX instructions are posted here: http://vicidial.org/docs/TILTX_SHAKEN_API.txt As for the container entry, the instructions say "place only the TILTX API key into the Container Entry", so you should place only the TILTX API key into the Container Entry, nothing else. The...

There has to be some kind of limit on the amount of mute time, or muted recordings where there had been some kind of issue with the agent session would be sitting there indefinitely and lost after a reboot. We chose 15 minutes because it seemed like a good balance between the allowed length of a...

Just email <support@vicidial.com> and mention this topic(with a link). As for adding a per-list CID Group Override, that would be more involved given the many sources and options for setting a CID on a call. I'd have to do a code review before I could quote you on that one.

With this dialplan entry, you are placing your phone calls twice. Doing this will result in inconsistent logging as well as the issues you are experiencing. If you need to use a failover carrier setup, I would recommend using OpenSIPs instead, since that is what it was designed to do on a...

1. in our tests with PauseMonitor, the results were inconsistent with how it worked in Meetme sessions, which is why we didn't add that feature to VICIdial earlier. 2. If the recording is muted for over 15 minutes, the recording file will be processed by the server and moved.

You would need to upgrade your system(which you should do anyway), since Asterisk 1.4 does not support that feature.

Our base rate is $200/hr, so $400 for 2 hours, then it's part of the VICIdial codebase.

The Agent API "switch_lead" function should return an "ERROR: agent_user does not have a live call..." response if you are trying to use it after hanging up the customer.

We finally had some time(and patience) to work on this over the last week and we've committed updated AST_VDadapt.pl code to svn/trunk revision 3480. Please upgrade and give it a try and let us know if you find any issues with it.

That specific error was fixed in svn/trunk a few days ago as I was trying to fix another issue.

Are there any errors on the web browser console when this happens?

I would consider these specific requests outside of the scope of VICIdial itself, since they don't directly involve VICIdial. I'm sure there are utilities somewhere that can do this already, even if they're just plugins to Nagios or iCinga.

Are you using the "Phone Only" page? (linked from Reports -> Admin Utilities) If so, and you are still have problems, post what shows up on the Asterisk CLI when you try to dial.

Apologies for not responding to this topic for 2 months, we've been pretty busy here. This looks like a good Feature Request for you to post on the Issue Tracker. It will require adding a new option in the Modify In-Group screen, and associated changes on the back-end. I will also add this to...

If you install Asterisk 16 on it, it is technically possible to make video calls through it if you configure it properly, but I wouldn't say it's ready for production, it's very inconsistent. We did a lot of testing of that last year and were disappointed in how bad it worked, failure to start...

You will probably have to raise the PHP variable limit to get to 2000+ CID Group entries, but other than that, it should work. For other clients with this kind of issue, we usually recommend just setting up a crontab entry to set the state to 'UF' for records with no populated state.

This has been fixed in svn/trunk revision 3464.

Yes, we have set this up for clients using the Phones' setting "Phone Context" and a corresponding Call Menu with the same Call Menu ID as the Phone Context, and then a Custom Dialplan Entry in that Call Menu with a PIN number entry portion before a call is allowed to go out. It's a bit complex...

There is often a database size tipping point after which problems start. We've seen this happen with many clients. I would suggest removing unused leads, setting up log archiving and running a database performance tuning script to see what my.cnf changes your system might benefit from.

We have been testing PJSIP, it does things very differently as far as the configuration files goes when compared to the old chan_SIP module. So, there will need to be some code adjustments on the VICIdial side to make it work with PJSIP phone and trunk configurations, but that is on our list to...