2 FreePBX boxes with conference call (how to reach each other)

Hi Experts,

I’m newbie to FreePBX and have setup PBX in flash and working over SIP trunk and can reach to both servers’ extensions.

I’m stuck in testing conference call. I setup conference number in server1 and it’s working as expected.

How do I configure to reach conference number in server1 from server2?

Is there any how to or guide me through.

Thanks,
John

Use an outbound route. So, on box 2 put in a route ( if the conf on box 1 is 7200) that has the pattern 72XX and send the call out that trunk.

Hi,

Thanks for your reply. I did create outbound route but still failed to get through. The error I received is "number you dial is not in service. Following are the console message capture in server1. I dialed from one of the extension from server2.

pbx*CLI>
– Accepting AUTHENTICATED call from 192.168.0.99:
> requested format = ulaw,
> requested prefs = (ulaw|alaw|gsm),
> actual format = ulaw,
> host prefs = (ulaw|alaw|gsm),
> priority = mine
– Executing [3000@from-trunk:1] Set(“IAX2/laptopuser-6052”, “__FROM_DID=3000”) in new stack
– Executing [3000@from-trunk:2] NoOp(“IAX2/laptopuser-6052”, “Received an unknown call with DID set to 3000”) in new stack
– Executing [3000@from-trunk:3] Goto(“IAX2/laptopuser-6052”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“IAX2/laptopuser-6052”, “”) in new stack
– Executing [s@from-trunk:3] Wait(“IAX2/laptopuser-6052”, “2”) in new stack
– Executing [s@from-trunk:4] Playback(“IAX2/laptopuser-6052”, “ss-noservice”) in new stack
– <IAX2/laptopuser-6052> Playing ‘ss-noservice.gsm’ (language ‘en’)
– Executing [s@from-trunk:5] SayAlpha(“IAX2/laptopuser-6052”, “3000”) in new stack
– <IAX2/laptopuser-6052> Playing ‘digits/3.gsm’ (language ‘en’)
– <IAX2/laptopuser-6052> Playing ‘digits/0.gsm’ (language ‘en’)
– <IAX2/laptopuser-6052> Playing ‘digits/0.gsm’ (language ‘en’)
– <IAX2/laptopuser-6052> Playing ‘digits/0.gsm’ (language ‘en’)
– Executing [s@from-trunk:6] Hangup(“IAX2/laptopuser-6052”, “”) in new stack
== Spawn extension (from-trunk, s, 6) exited non-zero on ‘IAX2/laptopuser-6052’
– Executing [h@from-trunk:1] Macro(“IAX2/laptopuser-6052”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“IAX2/laptopuser-6052”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“IAX2/laptopuser-6052”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“IAX2/laptopuser-6052”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“IAX2/laptopuser-6052”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘IAX2/laptopuser-6052’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘IAX2/laptopuser-6052’
– Hungup ‘IAX2/laptopuser-6052’

Hi,

After checking in Asterisk console message, I manage to add DID number in server1. Now, I can reach conference number in server1 from server2.

It would have been much easier if you had put the trunk in the “from-internal” context. That would have allowed it to access the internal dial plan and the conference.

Hi SkykingOH,

Many thanks for your reply. I understand more about context now. Adding context=from-internal solve all problems too. :slight_smile: