Hi Folks,
Strange one, I’m very much a newbie so could be missing something obvious here. I’m running FreePBX 2.7 on a VMware ESXi 4 Update 1 host server. This FreePBX instance is purely for Proof of Concept testing - it will be going on real hardware once we’re ready to deploy to production.
So, I’ve setup a test conference, with an Inbound Route of 8651 routing to this test conference instance. I’ve also setup a few test user extensions, 144, 133, etc, etc, with Inbound Routes setup to map to each. The PSTN trunk I’m using is an AudioCodes Mediant 1000 unit, inbound and outbound calls are fine via this trunk, I can send and receive calls to/from the PSTN from each user extn using X-Lite. The internal user extensions (133, 144, etc), are able to dial each other from X-Lite simply by dialling the 3-digit extension number. Voicemail is also working as expected.
When I dial into the test conference from an external source (for example, my mobile phone), the call is handled by the AudioCodes and FreePBX fine - I’m connected to the conference and others are able to join as expected. Seems OK so far.
However, when I try to dial the conference internally from the user extensions (I’m trying to dial simply 8651 from X-Lite), the call is immediately terminated, and the CLI displays the following:
[root@localhost asterisk]# asterisk -rvvvvvvvvv
Asterisk 1.4.24, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.
== Parsing ‘/etc/asterisk/asterisk.conf’: Found
Connected to Asterisk 1.4.24 currently running on localhost (pid = 3585)
Verbosity is at least 16
localhostCLI>
localhostCLI>
– Executing [8651@from-internal:1] ResetCDR(“SIP/144-09e027b0”, “”) in new stack
– Executing [8651@from-internal:2] NoCDR(“SIP/144-09e027b0”, “”) in new stack
– Executing [8651@from-internal:3] Progress(“SIP/144-09e027b0”, “”) in new stack
– Executing [8651@from-internal:4] Wait(“SIP/144-09e027b0”, “1”) in new stack
– Executing [8651@from-internal:5] Playback(“SIP/144-09e027b0”, “silence/1&cannot-complete-as-dialed&check-number-dial-again|noanswer”) in new stack
– <SIP/144-09e027b0> Playing ‘silence/1’ (language ‘gb’)
– <SIP/144-09e027b0> Playing ‘cannot-complete-as-dialed’ (language ‘gb’)
– <SIP/144-09e027b0> Playing ‘check-number-dial-again’ (language ‘gb’)
– Executing [8651@from-internal:6] Wait(“SIP/144-09e027b0”, “1”) in new stack
– Executing [8651@from-internal:7] Congestion(“SIP/144-09e027b0”, “20”) in new stack
== Spawn extension (from-internal, 8651, 7) exited non-zero on ‘SIP/144-09e027b0’
– Executing [h@from-internal:1] Macro(“SIP/144-09e027b0”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/144-09e027b0”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/144-09e027b0”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/144-09e027b0”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/144-09e027b0”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/144-09e027b0’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/144-09e027b0’
That’s all I have to go on… I know Meet-me is not recommended in VM-based environments, but as I said this is purely for PoC at this stage - the fact I can connect to this conference from an external source proves that the problem isn’t related to the VM environment, it must be something to do with my internal extension/routing configuration.
Let me know what further information you need from me. Any help greatly appreciated!
Thanks,
Alistair