Intermintent no audio

General Help
1

Hello,

I’ve been using freepbx for 6 months now for my shop phone and couldn’t be happier with its performance but for the last week I’ve been having intermittent no audio on one side of calls. Sometimes on incoming and other times on outgoing calls. I’d like to figure out where the problem lies so I can sort it out. Any advice? Called 2483453445 from freepbx 2484777200, no audio from 2483453445

[2013-11-30 13:15:23] VERBOSE[1727] pbx.c: – Executing [s@macro-dialout-trunk:22] Dial(“SIP/6-0000003b”, “SIP/FlowRoute/12483453445,300,”) in new stack
[2013-11-30 13:15:23] VERBOSE[1727] netsock2.c: == Using SIP RTP TOS bits 184
[2013-11-30 13:15:23] VERBOSE[1727] netsock2.c: == Using SIP RTP TOS bits 184
[2013-11-30 13:15:23] VERBOSE[1727] netsock2.c: == Using SIP RTP CoS mark 5
[2013-11-30 13:15:23] VERBOSE[1727] netsock2.c: == Using SIP RTP CoS mark 5
[2013-11-30 13:15:23] VERBOSE[1727] app_dial.c: – Called SIP/FlowRoute/12483453445
[2013-11-30 13:15:23] VERBOSE[1727] app_dial.c: – Called SIP/FlowRoute/12483453445
[2013-11-30 13:15:24] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is making progress passing it to SIP/6-0000003b
[2013-11-30 13:15:24] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is making progress passing it to SIP/6-0000003b
[2013-11-30 13:15:27] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is ringing
[2013-11-30 13:15:27] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is ringing
[2013-11-30 13:15:27] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is making progress passing it to SIP/6-0000003b
[2013-11-30 13:15:27] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c is making progress passing it to SIP/6-0000003b
[2013-11-30 13:15:29] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c answered SIP/6-0000003b
[2013-11-30 13:15:29] VERBOSE[1727] app_dial.c: – SIP/FlowRoute-0000003c answered SIP/6-0000003b
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/6-0000003b”, “hangupcall,”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [h@macro-dialout-trunk:1] Macro(“SIP/6-0000003b”, “hangupcall,”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/6-0000003b”, “1?theend”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:1] GotoIf(“SIP/6-0000003b”, “1?theend”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Goto (macro-hangupcall,s,3)
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/6-0000003b”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:3] ExecIf(“SIP/6-0000003b”, “0?Set(CDR(recordingfile)=)”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/6-0000003b”, “”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: – Executing [s@macro-hangupcall:4] Hangup(“SIP/6-0000003b”, “”) in new stack
[2013-11-30 13:15:43] VERBOSE[1727] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/6-0000003b’ in macro ‘hangupcall’
[2013-11-30 13:15:43] VERBOSE[1727] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/6-0000003b’ in macro ‘hangupcall’
[2013-11-30 13:15:43] VERBOSE[1727] features.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/6-0000003b’
[2013-11-30 13:15:43] VERBOSE[1727] features.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/6-0000003b’
[2013-11-30 13:15:43] VERBOSE[1727] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/6-0000003b’ in macro ‘dialout-trunk’
[2013-11-30 13:15:43] VERBOSE[1727] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on ‘SIP/6-0000003b’ in macro ‘dialout-trunk’
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: == Spawn extension (from-internal, 12483453445, 6) exited non-zero on ‘SIP/6-0000003b’
[2013-11-30 13:15:43] VERBOSE[1727] pbx.c: == Spawn extension (from-internal, 12483453445, 6) exited non-zero on ‘SIP/6-0000003b’

2

Your logs don’t contain any useful info to debug the issue, you can

rtp set debug on|ip

to see the rtp packets (audio) but whether they are coming through your router on the right port is not something asterisk can do, use tcpdump or wireshark to investigate.

(you have duplicate entries for the “full” log in the /etc/asterisk/logger* files)

3