vicidial.org

Hi
I am new at this and can't find the following info:

1. Kamailio: Used as a front entrance before Asterisk. Ok. But from where does it gets the configuration to route requests to Asterisk?
According to
https://goautodial.org/boards/1/topics/14249?r=15633
Kamailio is the one that gets the phones registrations...so...what is the purpose of Asterisk then?

2. Does anyone has a sip call flow I can see to proper understand which component is supposed to do what? Would be great for troubleshooting

Thanks in advance

kamailio doesn't handle audio, just signalling. Asterisk is a PBX, kamailio is a call routing system that does not handle the actual sound, excellent for billing.

have you looked at Vicibox.com's installation .iso? No kamailio. PDF installation instructions and then you switch to the Vicidial Manager's Manual. Goautodial is best used only for those who can not use the Vicibox .iso installer.

Hi,
Thanks for the answer.
Yeah, I am aware Kamailio is like a gateway only, however, that post made me even more confusing.
How does Kamailio says where to send calls, Asterisk in this case? By conf file correct?

I am installing in a EC2 instance, therefore, ISO files are not an option. I have installed the Vicidial in a local VMWare but as NAT is not doable there I have problems finding the correct configuration.

Regards

cavagnaro wrote:How does Kamailio says where to send calls, Asterisk in this case? By conf file correct?

cfg file. https://www.kamailio.org/w/2018/07/mini ... nfig-file/

cavagnaro wrote:I am installing in a EC2 instance, therefore, ISO files are not an option. I have installed the Vicidial in a local VMWare but as NAT is not doable there I have problems finding the correct configuration.

I don't know why you say NAT is "not doable". SIP doesn't appreciate NAT, especially for inbound calls to multiple servers, but it's entirely possible. And not really difficult if you only have one server.

Your best bet would be to skip kamailio entirely and just use Asterisk/Vicidial.

PS: If you try to run Vicidial in a Virtualized system, please keep your instance to only a few calls for testing. Note that when you try to "load it up" and run production, Virtual Vicidial will crash and burn when you exceed 3-4 calls or 1-2 agents (if you have a very powerful virtual server, you may exceed this ... but it's never guaranteed and generally expensive).

Well what I mean is that SIP by default doesn't support NAT. Software, of course, can fix this by configuration.
If I skip Kamailio, how to tell GoAUTODial to store users in Asterisk DB and not in Kamailio? Because right now it creates the users at Kamailio DB

sip does support nat. that's what the "externip" setting in sip.conf is for. That's why there's a NAT configuration option in the sip settings. SIP supports nat very well. It's when you have multiple servers trying to use one router that problems arise. If you forward UDP port 5060 and port range 10k-25k through the router to the asterisk server, nat is pretty much done (assuming the correct NAT yes/no setting for the sip account in question) AND either turn ON or OFF the "sip alg" in the router (some require it, some break with it in), sip eventually works.