Call routing from External PBX to trunk

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jprouty

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I have an asterisk PBX in my remote office which is not in my control, future change over to 3cx will take some time.
I would like to be able to allow the remote pbx to route calls to my external sip trunk.
on Asterisk this is easy as I can route to a trunk
Is this possible in 3cx? I have a sip trunk from that office to my 3cx working.
 
This is a good place to start...

https://www.3cx.com/docs/

It is going to come down to...sending the correct digits, and then the correct translation, and routing at the terminating PBX end.
 
The outbound trunk is on my 3cx instance
So call calling from the asterisk to the 3cx I am using a generic sip trunk the bridge never worked for me.
my remote pbx is japan so they want to connect to my pbx to make state side calls via my sip trunk.

I route the calls to my 3cx instance but how do I route inbound calls on that trunk to my pstn trunk
 
You won't be able to use a Generic SIP trunk as it cannot route traffic in the same manner a Bridge trunk. With a Bridge trunk, the other PBX will send digits that are recognised by the 3CX outbound rules, just as if a local extension had dialled them. As an example (no having any idea of the dial plans at either end). Remote extension dials 5914155551212. Remote PBX uses 5, or 59, to route via Bridge trunk to 3CX. Strip the 5, and send 914155551212 to 3CX. At the 3CX PBX, 914155551212 is in the outbound rules as a valid call.

You can leave the Generic SIP trunk in place, if some types of calls work over that. Use the Bridge trunk for calls direct to extensions, or outside calls.
 
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The odd part when I had the Bridge setup the calls never came to the 3CX instance.
I did tcpdump and did not see any traffic being sent.
I could make calls to the Asterisk no issues.
I can try again on the bridge setup.
THe documentation was good the issues were on my asterisk side.
 
bridge fails to authenticate

When I set the Master bridge


; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

on my asterisk system
Outgoing
host=IP address
port=5060
username=10007
fromuser=10007
secret=u5cwMj4ghA
sendrpid=yes
qualify=yes
type=peer


Incoming
host=192.168.149.130
port=5060
type=peer
trustrpid=yes
context=from-trunk

10007:[email protected]:5060/10007
 
That is just a random password I have rebuild the Master Bridge and used more of the sip debugging
I have all working as expected now.
Thank you for your assistance.
 
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