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So transfer to a call menu? That's not hard.

Are your DADHI drivers loading?

What method of login?

That is not how you use the Dial application. It should be: Dial(SIP/${EXTEN:0}@VoIPCarrier,,tTor) Fix this and try again. Then I would check the carrier log and the asterisk logs to check what you get back from your carrier when you try and dial.

You must reset your lists.

Are they inbound/closer calls? As they will be in vicidial_closer_log instead.

The '/2000' means you're registering on a specific extension at the carrier. If the trunk has multiple DIDs attached to it, you don't want to do this.

Yes, you should have the 'T' and 't' options in your dial command for your carrier. I don't know why the example SIP Carrier doesn't have it.

Also check if you have a general network problem. If you are using cheap switches this can cause all sorts of wierd problems. There's a program called 'mtr' that you can run on the server to check the connectivity to your agent PCs and the carrier(s). You want zero loss% and as little StDev...

If it's always the same agent, does that mean it's always the same agent pc? If so I'd check that first, plus its network connection. You see 'DISPO' and 'INCALL' in the log because only the lists get updated with the 'ERI' status. If you are getting a lot of those for this agent it suggests...

user = phone extension password = registration password domain/proxy = Server IP Note that if you are unsure what phones are being used, create a new one. You should document which workstations have which phone though (A spreadsheet plus labels plus computer hostnames is how I do it here).

Ah so you are using an E1/T1 card? As for looking at your channels (this is a ViciBox v5 telephony server, also works fine with v6. I am unsure if this command is correct for v3 etc). asterisk -rx'core show channels'shows you everything. I usually have about 100 channels open at any one time...

Yes, from what I understand, there is code to prevent 3WAY transfer conferences being used by agents logging in. The 'extension' field in the 'vicidial_conferences' table gets set to '3WAY_$user'. The Login checks for Nulls.

Yes that looks right. You should be able to switch carriers by changing your campaign dial prefix between 7 and 8.

In ViciDIAL go to Admin -> Phones. These are the accounts that Zopier uses.

I meant 'stuck channels' as in channels in asterisk that haven't hung up. Yes red is kernel in the htop CPU display. Another thought that occurred to me, how is your firewall setup? Do you allow SIP connections from anywhere on the Internet? If so it might be a brute force attempt on your SIP...

The dialplan for hera looks wrong.

You probably shouldn't post your carrier login username/password in public. Anyone can login and run up a phone bill in your name.

Did you have any stuck channels perhaps?

Yes, you don't need those.