Version 9: Can I now configure the remote with only...

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Iberocom

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I've seen that has come the new version and I still say that is excellent by parameters and features it has, but the question remains the same:

Can I configure the remote (softphone or any SIP-Phone "plugged" to an Internet connection) only setting the IP server, extension, ID and password, as I do with Asterisk? or I have to put a proxy / tunnel / stun for it work?

I know that comparisons are odious, but if you do not solve this problem, the problem remains is the same as that from the beginning. The asterisk will connect anything, anywhere in the world, and works in less than 1 minute. With 3CX I pulled the hair several weeks and in the end, I had to give up.

I guess the problem is Windows ... but if this is true ... the truth is that they could do 3CX for Linux!!!. It would be great ... ;-)
 
We have been using remote extensions since we started with 3cx.

With the 3cx soft phone on my laptop i put in our internal ip (for when I am in the office) and external for when i am roaming and with the new version i only have to tell it if i am in office or roaming.

I also have an SPA942 set up at home with no tunnel or vpn. I also have another PAP2T for two extensions in my home with no tunnel or vpn. All have worked with V8 and V9.

It sounds like you have a specific problem not the 3cx - so if you could give us more details we may be able to talk you through the set up.
 
Iberocom said:
Can I configure the remote (softphone or any SIP-Phone "plugged" to an Internet connection) only setting the IP server, extension, ID and password, as I do with Asterisk? or I have to put a proxy / tunnel / stun for it work?

You will need the IP Server, Extension ID and password, as you would with any SIP device, even on the local network. If your remote extension gets a public IP, then you won't need to set up STUN, however, if the remote extension is behind a router you will almost certainly have to set up STUN, The Proxy and Tunnel settings do not have to be used. The tunnel allows a connection through a single port, it's a tool to use if you wish.
 
Effectively Leejor. Today it is common to be behind a router ADSL, WiMax, Wi-Fi etc. no? I have a public fixed IP for my server and remote computers have a public variable IP in most cases. Place a proxy at each site with another 3CX on a computer, it is unjustifiable when I can do it with Asterisk without that.

I wonder if abc123 can plug your PAP2T at any Internet connection or a wi-fi, whether public or private, with or without router (in a hotel or airport, for example) and if it works.

Please understand me. I'm not saying either is better or worse. I say that if 1 + 1 = 2 I want 2, not 1. Everything can be improved and if it's possible, why not?
 
You don't need a 3CX Proxy Server running at a remote location you wish to use a 3CX extension. I have used both an ATA and VoIP WiFi phone at many public hotspots/hotels/friends homes, behind routers, with no issues (and, of course, no access to settings on the router). On occasion, if someone is choosing to block VoIP,a port conflict, or there is a proxy server in use, then it will fail.

You pretty much have to have the remote extension set to use STUN when there is a router at the remote location, which is most of the time.

The 3CX tunnel function (on the 3CX softphone) has come in handy at several airports where, for what ever reason, regular VoIP failed to work. A user would probably only run the 3CX proxy server, at a location, with a number of extensions (not just one), and only in tunnel mode if that option was required or desired.
 
Iberocom said:
Effectively Leejor. Today it is common to be behind a router ADSL, WiMax, Wi-Fi etc. no? I have a public fixed IP for my server and remote computers have a public variable IP in most cases. Place a proxy at each site with another 3CX on a computer, it is unjustifiable when I can do it with Asterisk without that.

I wonder if abc123 can plug your PAP2T at any Internet connection or a wi-fi, whether public or private, with or without router (in a hotel or airport, for example) and if it works.

Please understand me. I'm not saying either is better or worse. I say that if 1 + 1 = 2 I want 2, not 1. Everything can be improved and if it's possible, why not?

Just to back up what Leejor says - yes it can be done.

But I am not sure of your Asterisk comparison. Are you saying that you can connect any sip phone anywhere in the world (as your example above) and just enter the ip , userid and password and it will connect to the asterisk server, but not the 3cx server?

I dont see how that will be the case. Assuming you have correctly set up your routers etc on your end then the only difference would be the port that asterisk is listening on and the one that 3cx is listening on and usually they are the same.
 
They do not the slightest doubt that if I can change my asterisk by 3CX without touching anything on the remotes, I would do it immediately, but until version 8 I could not do it. I should find and pay technicians at each site to make the changes and try with STUN. but I do not like this solution, because that would depend on an outside server. With asterisk I do not need this, is independent of Nat, ports or other things. It is much more simple. Always works.

To anyone else this happens?

Thank you for your cooperation.
 
OK now I follow you.

This isnt going to be easy. Asterisk has the ability to be told whether to send back using the ports the incoming was received on or to use the information in the sip header (though I think this is an all or nothing setting?).

Even in some of those cases you still need a proxy.

I assume that you have remote clients set up using the nat setting in asterisk?

do you use a provisioning template to set up the remote phones? If so can you hijack that address to point to a new provisioning template? Or did you send them out already provisioned?

For example if the current extension has been set up for provisioning at http://<astrisk_server_provisioning>/<extension_template_location> then you can configure the extension in 3cx and it will save it at http://<3cx_server_provisioning>/<extension_template_location>. You can then either point your firewall/router to reroute that asterisk address to the 3cx address or copy the 3cx config to the astrisk location. Then the phone will pick it up and reconfigure itself with the tunnel if you set it up for that.

I would at least test it to see if it works.
 
I think that it is important to understand that at a minimum you will need the SIP ID, Password, Server IP. Then depending on if the remote phone is on a public IP address or NAT you need to enable stun. In the event that for some reason the firewall on the remote site gives you problem then consider the Tunnel (if softphone) tunnel proxy for hardware phones.
It can't get any easier than that... now if some hardware phone manufacturers would embrace the 3CX tunnel it would be nice
 
sigma1 said:
I think that it is important to understand that at a minimum you will need the SIP ID, Password, Server IP. Then depending on if the remote phone is on a public IP address or NAT you need to enable stun. In the event that for some reason the firewall on the remote site gives you problem then consider the Tunnel (if softphone) tunnel proxy for hardware phones.
It can't get any easier than that... now if some hardware phone manufacturers would embrace the 3CX tunnel it would be nice

I agree Charles. But what he is saying above is that he has remote extensions spread all over that he cannot get to.

Asterisk has a setting that doenst matter whether the remote firewall has NAT or not, you make the setting and it will switch from using the information in the sip header, to using the actual address and port it came in on.

3cx uses (the better in my opinion) the tunnel. But the OP needs to reconfigure those remote ones that get blocked, so they can use the tunnel.

So I suggested he use the config files to set up the tunnel for those remote ones, depending on how he has them provisioning with asterisk currently.
 
abc123 said:
Asterisk has a setting that doenst matter whether the remote firewall has NAT or not, you make the setting and it will switch from using the information in the sip header, to using the actual address and port it came in on.

Actually, we also have such setting.
 
Hi Archie,

Can you elaborate and give details regarding the setting you equate to the Asterisk NAT= parameter.

Thanks.
Eric
 
I don't know about Asterisk, but in 3CX there's custom parameter ALLOWSOURCEASOUTBOUND. Set it to 1 and PBX will reply to source address of INIVTE request.
 
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