Vicidial cluster mode with webrtc phone and carrier setup

[jayboo876]

Hi, I have vicibox 10 installation using downloaded iso and using the ViciBox_v9-install.pdf guide. The only addition I made was to upgrade asterisk to as16 using the vicibox-as16. I have a 5 server setup with db1 + (web1/archive) + web2 + (asterisk1/web) + (asterisk2/web). Note, I read post in this forum that indicated to install apache2 on asterisk servers if webrtc phones will be used.

However, I looking for some suggestions on best practices on distributing traffic across multiple servers;
- webrtc phones across the two (or more) asterisk/web servers
- web traffic the two (or more) web servers
- distributing carrier registration across multiple asterisk servers

I already If this is already covered in another post please add link here, or provide reference to the section in the admin manual manual that covers it. Looking for to any insight or suggestions.

Thanks

[carpenox]

Use the bulk phone insert link in the admin utilities to add phones to all your asterisk servers and create phone alias'

once you have phones on other dialers, they will start being used to dial from as well, using balanced dialing which you can set in "admin" then "servers"

Load balancing web servers can be done using njinx of which there is many articles on google on how to do so.

Hope this helps a bit

Chris

[jayboo876]

Use the bulk phone insert link in the admin utilities to add phones to all your asterisk servers and create phone alias'

once you have phones on other dialers, they will start being used to dial from as well, using balanced dialing which you can set in "admin" then "servers"

Load balancing web servers can be done using njinx of which there is many articles on google on how to do so.

Hope this helps a bit

Chris

Thanks Chris. This was really helpful. With regards to distributing traffic with my carrier, this is SIP only, and it seems I can only register carrier on a single asterisk. So am I correct to say inbound/outbound calls will terminate on same asterisk, or is there some other native config that will allow all asterisks to have the registration information?

Thanks

[carpenox]

Yea it will go thru iax tunnels

[williamconley]

Use the bulk phone insert link in the admin utilities to add phones to all your asterisk servers and create phone alias'

once you have phones on other dialers, they will start being used to dial from as well, using balanced dialing which you can set in "admin" then "servers"

Load balancing web servers can be done using njinx of which there is many articles on google on how to do so.

Hope this helps a bit

Chris

Thanks Chris. This was really helpful. With regards to distributing traffic with my carrier, this is SIP only, and it seems I can only register carrier on a single asterisk. So am I correct to say inbound/outbound calls will terminate on same asterisk, or is there some other native config that will allow all asterisks to have the registration information?

Thanks

Registration is for INBOUND only. By design, your carrier will send your inbound calls to either your hard-coded IP address OR your registered IP address, depending on how your carrier configures your DID. If you are using a registered SIP account for DID routing, the LAST asterisk server that successfully registered to the trunk will get the call. You may have multiple asterisk servers registering, and they will register at timed intervals (which are controlled by settings in sip.conf or sip-vicidial.conf).

None of which is related to outbound dialing. Outbound dialing is permitted by the carrier through hard-coded IP address authorization (in most cases) and may also be permitted through User/Password authentication as well (although there are a few Token-based carrier accounts, that merely a variation of user/pass IMHO).

So: Registration or IP for Inbound, which may rotate if you have multiple servers registering at intervals and user/pass and/or IP for outbound.

Also worthy of note: MOST carriers allow for a "hunt/failover" system of some sort, where you have multiple asterisk servers and the availability of the asterisk server(s) determine where the inbound calls route. This is so that a down server doesn't stop traffic flow, you have multiple servers so they would fail over to the next server in the hunt. If you talk to your carrier, they should have configuration options available for this (unless they are tiny or newly created). This method of failover and/or balancing is designed to allow control of the call flow and avoid the random nature of mutliple asterisk servers registering to a single sip account.

[jayboo876]

@williamconley, Thanks for the detailed explanation. I now have a good understanding of the inbound/outbound call flows with multiple asterisks.

Thanks