Agent Phone not Calling on Login - Cloud Server - No CLI

[daryn]

Hi,
So I have been mucking around for quite some time and managed to get asterisk and Vicidial semi-operational on a google cloud server. It is running Centos 6.6.
Asterisk 1.8.23.0-vici currently running on vicidial (pid = 1778)

When I log in to the agent webpage, the agent phone does not ring. There is also no information populated in the CLI when an agent logs in. I am quite stumped as to how to go about solving this. The phones work. They are connected and working. The only problem there (to solve later) is I cannot dial between them. I get "the person at.... is unavailable" message.

One oddity which might be a clue, is if I type asterisk -r I get the following: Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) But then if I go to cd /etc/asterisk and type asterisk, I can then type asterisk -r and the CLI will display. I am presuming there was some error upon install. I have gone through and reinstalled it though, but I get the same error. Maybe it's the startup config pointing to the wrong folder?

The screen -ls output is:
1809.ASTVDremote (Detached)
1818.ASTconf3way (Detached)
1799.ASTsend (Detached)
1803.ASTlisten (Detached)
1806.ASTVDauto (Detached)
1797.ASTupdate (Detached)
1633.asterisk (Detached)
1628.astshell20150603024128 (Detached)
1815.ASTfastlog (Detached)
1812.ASTVDadapt (Detached)
10 Sockets in /var/run/screen/S-root.

Would love some help to try and get this sorted.

One other idea I thought is the network configuration and settings. Maybe that is a clue? I'm not really sure how to set that all up or if it is configured correctly. I have tried mucking around in there but nothing seems to make any difference.

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) You should build this locally first. Install it from the Vicibox.com .iso image. Use the PDF from Vicibox.com to perform the installation, then use the PDF of the Vicidial Manager's Manual (free version from eflo.net!) to configure the server for use. Step By Step instructions are included. Free. Then (when you've got that running in a vSphere or VirtualBox or physical machine), you may be able to finagle a running server in the cloud.

4) DO NOT run Vicidial in the cloud. LOL (It'll eat all your time, it'll never work, and you'll have a great education in why when you're done ... but a lot of time will have passed that could have been useful). Vicidial does not run Virtual for more than 1 or 2 agents (great for testing, not so much for production ... unless your production is for two agents!)

5) IF your server is not aware of it's identity, it will not dial the phone it has been ordered to dial. IE: If /etc/astguiclient.conf does not contain the IP of this server which must match the "admin->phones" Phone's server IP ... then it will not be aware it has been instructed to dial. In the conf file, the "this server's IP" variable is relevant. In the admin->phones entry for the phone being logged in, that same IP must appear. So it knows it's identity, and that the phone is assigned to it.

6) Happy Hunting!