SIP calls between 3CX and other IP PBX over WAN

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urtho

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Hi,

Have simple setup here. Two branches linked with VPN.
One branch has Asterisk the other one has 3CX.
Each IP PBX has it's own local SIP trunks to ITSPs.

Goal one :
have proper (internal) caller id when calling from 3CX to Astrisk over WAN.

Goal two:
have RTP streams go directly between phones for interoffice calls.

PBXes and phones are in different private subnets reachable via default gateway.

Is 1 & 2 doable on v9 3CX ?
My current setup proxies RTP streams and puts external IP in SDP so I have no audio.
Also all caller-ids are spoofed to a default SIP Trunk Caller ID when going from 3CX to Asterisk.

Urtho,
 
I will prefix this by saying I have little or no experience with Asterisk.

3CX has a bridge mode where it can be a master or slave so if Asterisk can do the same then that is all you need.

You can have multiple bridges in 3cx and each is treated as a voip provider in the system (in laymans terms) meaning you can make calls through it. Use an outbound rule at each end to tell it to use the bridge (e.g. dial 11<ext> to go out through the bridge and strip the 11 prefix).

That way you "should" see the extension number of the caller (but this does depend on any config changes you may have made to the sip headers etc either end).
 
So the bridge thing is not limited to "3CX to 3CX" scenario but one can have 3CX to AN IP PBX bridge ?
 
Hi Urtho,

The subject is about WAN. The post is about VPN. WAN and VPN are slightly different things, right?
Could you please describe the topology of your network in detail?

3CX-to-3CX bridge is not (always) a "VoIP trunk". So, if you had no success using basic method of integration (using VoIP trunk) then it (the bridge) could not help you to resolve the issue because the real problem may be the network topology.

Thanks
 
Each office has a private subnet and both are routed over VPN without any kind of firewalling.

To make things simpler, lets assume I have 3CX at both offices.

Using 3CX bridge I am thus far unable to:
1) get direct RTP between phones across locations - 3CX proxies RTP streams. Not the thing I am used to in Asterisk, sipXecs, ... Am missing a way to list all my "local" subnets in 3CX.

2) get proper internal caller-id working. All calls have the caller ID set to 3CX bridge user. I can use outbound caller-id to override it but then I loose the ability to use the setting for outside (PSTN) calls - and rewriting all caller-ids globally @ gateway level is not what I am looking for. Am missing a way to allow internal caller-ids show up in 3CX bridge calls .

Urtho,
 
urtho said:
Each office has a private subnet and both are routed over VPN without any kind of firewalling.

To make things simpler, lets assume I have 3CX at both offices.

Using 3CX bridge I am thus far unable to:
1) get direct RTP between phones across locations - 3CX proxies RTP streams. Not the thing I am used to in Asterisk, sipXecs, ... Am missing a way to list all my "local" subnets in 3CX.

2) get proper internal caller-id working. All calls have the caller ID set to 3CX bridge user. I can use outbound caller-id to override it but then I loose the ability to use the setting for outside (PSTN) calls - and rewriting all caller-ids globally @ gateway level is not what I am looking for. Am missing a way to allow internal caller-ids show up in 3CX bridge calls .

Urtho,

No, bridging doesnt go direct from phone to phone because a lot of bridges are not via vpns. I agree you may want this so i would suggest adding it as a feature request at http://3cx.ideascale.com

However it does allow sip dialing so you can call ext@ipaddress and your routing will connect phone to phone that way.

Are you sure about #2? I am pretty sure (but will have to wait to get in the office on monday to confirm) that I see the caller extension number on my bridges when they call.
 
You could also try using the 3CX tunnel feature on the outside of your site-to-site VPN - I had a situation with a V8 setup where direct UDP across an IPSEC VPN tunnel wasn't working (I later realised that the client had setup their routing badly, but that's another discussion, I guess) so I tried using the tunnel instead to establish the bridge - worked a treat and made it slightly easier to setup QoS for the interoffice calls.

3CX to Asterisk - best idea is to get a second 3CX Server, otherwise I would try SIP calling as abc123 suggested.
 
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