Running FreePBX 2.9 loaded from ISO obtained here. core show version sows this:
Asterisk 1.8.5.0 built by root @ freepbxdev1.schmoozecom.net on a i686 running Linux on 2011-07-12 02:47:30 UTC
My Polycom phones are VPN connected to a SIP-based Rhino box at a local colo. The box sits behind a Cisco ASA5509 with a 1:1 NAT.
When I dial the conference bridge extension DID xxx-xxx-5201 I get silence. /var/log/asterisk/full shows that the call is going through, that the system is playing the prompts (but I hear nothing on the phone), and that the system is receiving my PIN. But then I get “User entered nothing”. I can get to the bridge if I dial 5202 from the phone, and if I dial from a cell into the bridge. At times even dialing the 4-digit extension of the bridge does not work, so the condition is reversed. It is random.
here is the content of /var/log/asterisk/full from right before the DFMF entry through disco. You can see that I enter the PIN, but the system does not hear it, nor am I hearing the system say anything… Thanks for any help.
[2012-01-26 14:15:43] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:8] Set(“SIP/VI-00000cc4”, “PINCOUNT=0”) in new stack
[2012-01-26 14:15:43] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:9] Read(“SIP/VI-00000cc4”, “PIN,enter-conf-pin-number,”) in new stack
[2012-01-26 14:15:43] VERBOSE[315] file.c: – <SIP/VI-00000cc4> Playing ‘enter-conf-pin-number.ulaw’ (language ‘en’)
[2012-01-26 14:15:47] DTMF[312] channel.c: DTMF begin ‘1’ received on SIP/7272-00000cc0
[2012-01-26 14:15:47] DTMF[312] channel.c: DTMF begin passthrough ‘1’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end ‘1’ received on SIP/7272-00000cc0, duration 300 ms
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end accepted with begin ‘1’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end passthrough ‘1’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF begin ‘5’ received on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF begin passthrough ‘5’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end ‘5’ received on SIP/7272-00000cc0, duration 320 ms
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end accepted with begin ‘5’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF end passthrough ‘5’ on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF begin ‘2’ received on SIP/7272-00000cc0
[2012-01-26 14:15:48] DTMF[312] channel.c: DTMF begin passthrough ‘2’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end ‘2’ received on SIP/7272-00000cc0, duration 320 ms
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end accepted with begin ‘2’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end passthrough ‘2’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF begin ‘3’ received on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF begin passthrough ‘3’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end ‘3’ received on SIP/7272-00000cc0, duration 320 ms
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end accepted with begin ‘3’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF end passthrough ‘3’ on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF begin ‘#’ received on SIP/7272-00000cc0
[2012-01-26 14:15:49] DTMF[312] channel.c: DTMF begin passthrough ‘#’ on SIP/7272-00000cc0
[2012-01-26 14:15:50] DTMF[312] channel.c: DTMF end ‘#’ received on SIP/7272-00000cc0, duration 420 ms
[2012-01-26 14:15:50] DTMF[312] channel.c: DTMF end accepted with begin ‘#’ on SIP/7272-00000cc0
[2012-01-26 14:15:50] DTMF[312] channel.c: DTMF end passthrough ‘#’ on SIP/7272-00000cc0
[2012-01-26 14:15:56] VERBOSE[315] app_read.c: – User entered nothing.
[2012-01-26 14:15:56] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:10] GotoIf(“SIP/VI-00000cc4”, “0?USER”) in new stack
[2012-01-26 14:15:56] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:11] GotoIf(“SIP/VI-00000cc4”, “0?ADMIN”) in new stack
[2012-01-26 14:15:56] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:12] Set(“SIP/VI-00000cc4”, “PINCOUNT=1”) in new stack
[2012-01-26 14:15:56] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:13] GotoIf(“SIP/VI-00000cc4”, “0?h”) in new stack
[2012-01-26 14:15:56] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:14] Playback(“SIP/VI-00000cc4”, “conf-invalidpin”) in new stack
[2012-01-26 14:15:56] VERBOSE[315] file.c: – <SIP/VI-00000cc4> Playing ‘conf-invalidpin.ulaw’ (language ‘en’)
[2012-01-26 14:15:59] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:15] Goto(“SIP/VI-00000cc4”, “READPIN”) in new stack
[2012-01-26 14:15:59] VERBOSE[315] pbx.c: – Goto (ext-meetme,5201,9)
[2012-01-26 14:15:59] VERBOSE[315] pbx.c: – Executing [5201@ext-meetme:9] Read(“SIP/VI-00000cc4”, “PIN,enter-conf-pin-number,”) in new stack
[2012-01-26 14:15:59] VERBOSE[315] file.c: – <SIP/VI-00000cc4> Playing ‘enter-conf-pin-number.ulaw’ (language ‘en’)
[2012-01-26 14:16:12] NOTICE[3368] chan_sip.c: Disconnecting call ‘SIP/VI-00000cc4’ for lack of RTP activity in 31 seconds
[2012-01-26 14:16:12] NOTICE[3368] chan_sip.c: Disconnecting call ‘SIP/VI-00000cc2’ for lack of RTP activity in 31 seconds