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
muyousif wrote:1. Do I need to create all sip phones accounts on both servers?
"Need"? No. But should you ... yes. Use the bulk phone adder tool. It allows creating load balancing phone entries across all servers! When you use it, do not overlap any existing phone entires. Start fresh.
muyousif wrote:2. Sip phones need to be register to both servers or need to register to first primary server?
Sip phones only need to register to one server, it matters not which. As long as that is the phone account to which the agent logs in. But see the answer above for #1, agents should be using the alias to log in for a load balanced system.
muyousif wrote:3. If supervisor or manager's extension is registered on first server then how he will be able to agent registered to second server?
I assume you meant "how will he be able to Monitor". Vicidial's real time screen monitor method handles routing of the monitor call to the manager's phone. Read the manual.
muyousif wrote:4. If DID is routed to first telephony server then how system will route it to the agent / sip phone registered to second server.
1) DID is routed to server (by Telco interface directly, or through registration which alllows Vicidial to reach out and "grab" an account from a Telco).
2) Server routes call to "trunkinbound" extension context upon arrival
3) "trunkinbound" routes call to appropriate AGI script (turning control over to Vicidial (IE: Matt Florell's Monster)
4) Vicidial routes the call according to how you set this DID to route ... USUALLY to an Ingroup
5) Ingroup routes the call to the Meetme Conference of the agent that the system deems "next". IF Necessary, this will include bouncing the call to a different dialer! Automagically!
Note that if you do not route the call to an Ingroup, this may or may not hold true.
Hopefully this will dispel some of the Magic. To dispel more of it: Watch the Asterisk CLI of both servers. You'll see the call arrive, the agi perl scripts fire, the agi scripts executing more extensions as necessary to route the call to the agent (through an IAX inter-server trunk if required
). Thanks Matt.
muyousif wrote:5. My first telephony server is also my archive server so how recorded calls of on second server will be copied to archive server?
Check crontab -e of each dialer and activate the recording FTP script.
Test it with a single recording using --debugX before fully activating it .. be sure the new recording link works on the lead record of the file transferred .. it's a pain to fix them later if a bunch have the wrong link!
. Now going to increase number of agents so want to add new telephony server. I have some queries given below