So SNG7 has problem with NIC on VMWare and doesn't detect any network adapter?
Because I install FreePBX 14 from .iso, and it doesn't detect NIC too.
I hope Sangoma guys solve this problem as soon as possible.
Regards.
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How to Upgrade FreePBX Distro 13 to 14
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How to Upgrade FreePBX Distro 13 to 14
The problem I had was that when upgrading from 13 to 14, it would not detect internet connection, exactly as your screenshot.
I just gave up, and downloaded the .iso and installed fresh in to my vmware 6.5 VM, configured netcard driver as NE1000. Works fine now.
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Recording Access
Thanks for reply dickson.
Any free solution available for this ?
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What is the best way to connect multiple FreePBX together?
Try what? What is laid out in that blog post? Of course but again as I pointed out before that blog post is old and well, incomplete. It shows how to send CALLS to/from Asterisk. What about the rest? Is this a Proxy setup, is it a Upper Registrar setup? Nothing in this post talks about ANYTHING but calls.
How are REGISTERs handled? SUBSCRIBEs? OPTIONs? Who is doing the keep-alive checks on the endpoint to make sure it's still alive and where it is at? How are other things handled like VoiceMail since this setup sends calls at a 50/50 weight to the TWO Asterisk servers for making calls.
Nothing in this blog post covers any of the other RELEVANT parts of handling the users and their accounts. Where does it send the calls when the users dials *97 or *98? Where does it send calls if the user is BUSY or returns a NO ANSWER? How does it get to VoiceMail? Where is the VoiceMail route?
So this is nothing more than a PIECE of a puzzle when it comes to handling VoIP services. The piece that is handling making/receiving calls. Those other annoying things like who handles the Registrar/Location services, where things like VoiceMail are handled or any other important parts of the puzzle are missing.
And even more so, as I pointed out before this doesn't account for FREEPBX! These blogs posts account for ASTERISK which works under the assumption you are free to control and edit the dialplan and configs freely without restriction.
@dicko I'm pretty confident I'm running Kamailio + FreePBX in ways you have yet to touch or considered.
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What is the best way to connect multiple FreePBX together?
Great, so can you offer any help then?
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How to distinguist between trunks on incoming calls?
Trunks are not determined by alphabetical order, just a top down order. You can order your peer configs anyway you want in sip.conf. As well "Inbound Routes" is not an Asterisk thing, it's a FreePBX/common name thing.
How a trunk is matches is based on how that trunk is setup. If you have two trunks setup with type=PEER and they have the same host= setting, the FIRST one in the sip.conf that matches those TWO settings, wins. Now a Friend/User type will care about the username portion and want to match on that over IP.
Trunks do two things for Asterisk. 1) They determine if the source of the call/request is actually allowed to send to Asterisk. This is the PEER vs FRIEND/USER part. 2) They tell matches on that Trunk to send incoming calls to a specific context in the dialplan. This would be your "Inbound Routes" section.
Please do not confuse the FreePBX implementation of Asterisk for the "normal model" of Asterisk. Because the "normal model" of Asterisk is configure it how you want. Which would mean person A and B's "model" will be different.
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What is the best way to connect multiple FreePBX together?
@AIC2000 I can but you need to determine a path you're going to take first. Kamailio will do what you want. It's how I manage my peers to carriers and track how many inbound/outbound/overall calls are in use with them. That requires the dialog management module to be in place and logic written for that.
You are going to need to determine how much FreePBX will be doing over all vs how much the proxy will be doing. Will you want to use the proxy in a "pass-thru" mode which sends the requests straight to the PBX and let's the PBX deal with everything or will you want it in Upper Registration mode (takes way more to do) but then everything registers/talks to the Kamailio system before (and only when needed) sending requests to the PBX.
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Client friendly CDR's
That is because you are confusing an ADMIN tool for looking a call details in depth vs an actual reporting tool. You're also confusing a CDR with Call Event Logging. A CDR tells you about the call, if it was Answered/BUSY/etc, how long it was, who the caller and callee's are. Simple stuff.
Stuff like was this call put on HOLD or was it PARKED, who picked it up after it was PARKED, etc, etc, etc. Those are call EVENTS that happen during a call. CDR and CEL are two different things.
So a CDR reporting tool may show you Call X came in at Y time and Z disposition and how long the call was. But if you want to know all the things that happened during the call you need to look at the Call Event Logs which is an entirely different database.
The real answer is, if you want a reporting tool that covers all those things you are either paying for it from a third party somewhere or you're writing your own.
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What is the best way to connect multiple FreePBX together?
What would be the advantage of doing it eithe way?
I want each business to have their own FreePBX setup but want to pool outbound sip channels between multiple FreePBX while maintaining caller IDs
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Client friendly CDR's
The issue that I'm having is that the CDR tool in FreePBX offers TOO much information. She needs something simple.
Can anyone suggest a 3rd party tool for simple reporting that an Office Manager can use to see calls for the entire company?
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What is the best way to connect multiple FreePBX together?
Well, you put Kamailio "between" your FreePBX systems and your providers. So Kamailio peers with the providers so that incoming calls hit Kamailio and use on of the routing modules to send those calls to the proper FreePBX, each that would peer with Kamailio.
Then for outbound you just have a trunk between FreePBX and Kamailio that sends to call to Kamailio which does the reverse for the routing, sending the call out to the carrier of your choice.
CallerID will be intact for outbound calls from FreePBX since you're not messing with them at the Kamailio level.
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Client friendly CDR's
AsterNIC CDRs
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What is the best way to connect multiple FreePBX together?
Yes thats what I need to do, I just don't know where to start. Is this achievable via a gui?
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What is the best way to connect multiple FreePBX together?
P.S. I know how to do the Asterisk / FreePBX side of things
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What is the best way to connect multiple FreePBX together?
There is no GUI for Kamailio. You have to code this by hand.
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Phone does not get MWI from external voicemail server
You need to tell your phones to subscribe to FreePBX-B for voicemail. By default phones will use the host/proxy for the SIP account if the voicemail subscription details are left empty.
So right now your phones are subscribing to FreePBX-A for voicemail MWI because that's default.
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What is the best way to connect multiple FreePBX together?
Whats this then?
http://siremis.asipto.com/
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What is the best way to connect multiple FreePBX together?
That is a backend database management and reporting GUI. It will let you run some commands from the GUI but as far as the actual call routing logic, it does nothing.
Want to add SIP accounts, it does that. Want to add DIDs to the routing tables, does that, Domains, etc. It lets you add/delete/modify information in the database that your routing logic will use as a data source.
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SIP Trunk Providers
Need some recommendations or suggestions on SIP providers for a single office with 5 DIDs and average 25-30 calls per 8 hour day. First time setting up FreePBX and we're at the stage of ordering SIP service and activating it. I've been reading materials from Twilio, Flowroute, Nextiva, and SIPStation. I think for a small shop like ours, the pay as you go model might be the best. Any recommendations are greatly appreciated.
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What is the best way to connect multiple FreePBX together?
Right so thats no good then. Couldn't I use different siremis extensions to map to different extensions at my sip supplier?
Is there any documented examples or guides on how this simple setup can be achieved or is it a case of suck it n' see?
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