Connecting two FreePBX boxes with a SIP Trunk

General Help
1

Hello

I have Server A and Server B. I want to configure a SIP trunk between the two servers so that all outbound calls on Server A route through Server B which is connected to my carrier.

I’ve got a SIP trunk registered between them both, but when I attempt a call, it never hits Server B. Server A provides the following output:

– Executing [s@macro-dialout-trunk-predial-hook:1] SIPAddHeader(“SIP/652-00001aec”, “P-Asserted-Identity:sip:[email protected]”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:2] MacroExit(“SIP/652-00001aec”, “”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/652-00001aec”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:19] ExecIf(“SIP/652-00001aec”, “1?Set(CONNECTEDLINE(num,i)=2146749682)”) in new stack
– Executing [s@macro-dialout-trunk:20] ExecIf(“SIP/652-00001aec”, “1?Set(CONNECTEDLINE(name,i)=CID:4698503805)”) in new stack
– Executing [s@macro-dialout-trunk:21] GotoIf(“SIP/652-00001aec”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:22] Dial(“SIP/652-00001aec”, “SIP/Call-Router-A/2146749682,300,Tt”) in new stack
– Executing [s@macro-dialout-trunk:23] NoOp(“SIP/652-00001aec”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21”) in new stack
– Executing [s@macro-dialout-trunk:24] GotoIf(“SIP/652-00001aec”, “0?continue,1:s-CONGESTION,1”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/652-00001aec”, “RC=21”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/652-00001aec”, “21,1”) in new stack
– Executing [21@macro-dialout-trunk:1] Goto(“SIP/652-00001aec”, “continue,1”) in new stack
– Executing [continue@macro-dialout-trunk:1] NoOp(“SIP/652-00001aec”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:2] Set(“SIP/652-00001aec”, “CALLERID(number)=652”) in new stack
– Executing [2146749682@restrictedroute-9bf31c7ff062936a96d3c8bd1f8f2ff3:7] Macro(“SIP/652-00001aec”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/652-00001aec”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/652-00001aec”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/652-00001aec”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/652-00001aec”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/652-00001aec> Playing ‘all-circuits-busy-now.ulaw’ (language ‘en’)
– <SIP/652-00001aec> Playing ‘pls-try-call-later.ulaw’ (language ‘en’)

Here are my trunk config’s for both servers…

Server A
type=friend
context=from-internal
qualify=yes
host=IP ADDRESS OF SERVER B
insecure=invite
allow=all

Server B
type=friend
context=from-internal
qualify=yes
insecure=port,invite
host=CARRIER IP
fromdomain=CARRIER IP
disallow=all
allow=g729,ulaw

Can someone please help me understand what is wrong with my trunk config?

Thank you so much!

2

Where’s the trunk from/to Server A on Server B?

3

Oops, I have added that additional trunk.

Now the call is hitting Server B, but getting a "the number is not in service prompt:

[2017-08-31 14:25:30] WARNING[20254][C-0000002d]: Ext. s:3 @ from-trunk: Friendly Scanner from IPADDRESS OF SERVER A
– Executing [s@from-trunk:4] Wait(“SIP/call-router-a-00000014”, “2”) in new stack
– Executing [s@from-trunk:5] Playback(“SIP/call-router-a-00000014”, “ss-noservice”) in new stack

I can register an extension on Server B and place an outbound call fine, so the trunk there appears to be working.

4

I’ve got it working… I had to create an ANY/ANY inbound route and point that to my carrier trunk on Server B.

Sorry for the stupid question, I haven’t linked two PBX’s together with a SIP trunk in the past.

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5

I do have one question.

When the call from Server A processes through Server B, the outbound caller ID from Server A is lost and showing up as unknown on the receiving end.

Any idea how to correct this?

6

I’d start with Googling trustrpid and “Intracompany route” and search around in that little universe. One of those will lead you to a solution in your problem set.

8

I believe I have resolved this.

I added the following to Server A trunk PEER Details: sendrpid=yes
I added the following to Server B trunk PEER Details: trustrpid=yes

I also set the outbound routes on Server B as intra-company routes.

Caller ID is now passing through from Server A, to Server B, to the receiving party.

Thanks for your help Dave.

9

Actually… this passes the caller ID through, but on Server B CDR reports, I’m unable to see which trunk that calls are coming in on, so I have no idea which trunk the call is originating from, which is information I need to know.

Is there a better way to do this?

10

I would have used an IAX2 link between PBXes.
Only one UDP port to open and manage on your firewall.
I have many PBXes federated that way.

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