Hi everybody,
I have a problem with my new FreePBX 2.8.1.4 installation and the Conferences setup.
I created some test-conferences (6099 in this case) and tried to call them with unregistered SIP extensions. When I do that I get the following command lines:
– Executing [6099@from-sip-external:1] NoOp(“SIP/192.168.0.147-0000000d”, “Received incoming SIP connection from unknown peer to 6099”) in new stack
– Executing [6099@from-sip-external:2] Set(“SIP/192.168.0.147-0000000d”, “DID=6099”) in new stack
– Executing [6099@from-sip-external:3] Goto(“SIP/192.168.0.147-0000000d”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/192.168.0.147-0000000d”, “1?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] GotoIf(“SIP/192.168.0.147-0000000d”, “1?setlanguage:from-trunk,6099,1”) in new stack
– Goto (from-sip-external,s,3)
– Executing [s@from-sip-external:3] Set(“SIP/192.168.0.147-0000000d”, “CHANNEL(language)=de”) in new stack
– Executing [s@from-sip-external:4] Goto(“SIP/192.168.0.147-0000000d”, “from-trunk,6099,1”) in new stack
– Goto (from-trunk,6099,1)
– Executing [6099@from-trunk:1] Set(“SIP/192.168.0.147-0000000d”, “__FROM_DID=6099”) in new stack
– Executing [6099@from-trunk:2] NoOp(“SIP/192.168.0.147-0000000d”, “Received an unknown call with DID set to 6099”) in new stack
– Executing [6099@from-trunk:3] Goto(“SIP/192.168.0.147-0000000d”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/192.168.0.147-0000000d”, “”) in new stack
– Executing [s@from-trunk:3] Wait(“SIP/192.168.0.147-0000000d”, “2”) in new stack
– Executing [s@from-trunk:4] Playback(“SIP/192.168.0.147-0000000d”, “ss-noservice”) in new stack
– <SIP/192.168.0.147-0000000d> Playing ‘ss-noservice.ulaw’ (language ‘de’)
– Executing [s@from-trunk:5] SayAlpha(“SIP/192.168.0.147-0000000d”, “6099”) in new stack
– <SIP/192.168.0.147-0000000d> Playing ‘digits/6.ulaw’ (language ‘de’)
– <SIP/192.168.0.147-0000000d> Playing ‘digits/0.ulaw’ (language ‘de’)
– <SIP/192.168.0.147-0000000d> Playing ‘digits/9.ulaw’ (language ‘de’)
– <SIP/192.168.0.147-0000000d> Playing ‘digits/9.ulaw’ (language ‘de’)
– Executing [s@from-trunk:6] Hangup(“SIP/192.168.0.147-0000000d”, “”) in new stack
== Spawn extension (from-trunk, s, 6) exited non-zero on ‘SIP/192.168.0.147-0000000d’
– Executing [h@from-trunk:1] Hangup(“SIP/192.168.0.147-0000000d”, “”) in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on ‘SIP/192.168.0.147-0000000d’
The redirect to the conference only works with registered SIP extensions, even though I enabled anonymous incoming calls.
This is the command line with a registered extension:
-- Executing [6099@from-internal:1] Answer("SIP/9141-0000000f", "") in new stack
-- Executing [6099@from-internal:2] Wait("SIP/9141-0000000f", "1") in new stack
-- Executing [6099@from-internal:3] Playback("SIP/9141-0000000f", "conf-thereare") in new stack
-- <SIP/9141-0000000f> Playing 'conf-thereare.ulaw' (language 'de')
-- Executing [6099@from-internal:4] MeetMeCount("SIP/9141-0000000f", "6099") in new stack
== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Parsing ‘/etc/asterisk/meetme_additional.conf’: == Found
– <SIP/9141-0000000f> Playing ‘digits/0.ulaw’ (language ‘de’)
– Executing [6099@from-internal:5] MeetMe(“SIP/9141-0000000f”, “6099,cMs1”) in new stack
== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Parsing ‘/etc/asterisk/meetme_additional.conf’: == Found
– Created MeetMe conference 1023 for conference ‘6099’
– Started music on hold, class ‘default’, on SIP/9141-0000000f
– Stopped music on hold on SIP/9141-0000000f
– Started music on hold, class ‘default’, on SIP/9141-0000000f
In fact I don’t want to register any phone, because the FreePBX system is only used for conferencing and behind an additional phone system and a firewall. So I’d like the system to handle any inbound call like the above.
On my old asterisk system this worked very fine. For the future a web interface like FreePBX is necessary, because my colleagues have no experiences with Asterisk on a Linux console…
Thank you in advance!