search for: app_mixmonitor

Hello All, I posted a bug on the 14th of this month, and haven't heard anything back. However, I've since discovered that the problem is not in chan_iax.c as I originally thought, it's actually app_mixmonitor.c. Basically when I use 1.4.26.2 with an ilbc codec between two asterisk servers trunked via IAX, with mixmonitor Asterisk crashes on me. Here's a link to the post: https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=16070. Can someone possibly assign it to the right application. I s...

...ahdi.c: -- Channel 0/6, span 7 got hangup, cause 27 [Mar 18 17:04:15] VERBOSE[27353] pbx.c: == Spawn extension (incoming-pri, sw-30-218543080, 11) exited non-zero on 'DAHDI/189-1' [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: q931_hangup: other hangup [Mar 18 17:04:15] VERBOSE[27354] app_mixmonitor.c: == MixMonitor close filestream [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, hold-state Idle [Mar 18 17:04:15] VERBOSE[27353] chan_dahdi.c: NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null, hold-state Id...

...USB disk fails for some odd reason, like hardware failure or power failure..., asterisk complains that it is unable to write the recording to the USB Drive either by crashing asterisk or generate an infinite loop of errors on the asterisk console (Input/Ouptut Errors). If I try to unload the module app_mixmonitor.so, asterisk crashes. I am wondering if we can make asterisk stop recording on all the recorded calls and not to crash/generate errors if it does not see the USB drive any more. i thought the easiest way is to unload the app_mixmonitor module, but unfortunately it is crashing asterisk at the same...

...exten => _911.,9,Dial(${TRUNK}/${EXTEN:3},,To) exten => _911.,10,Hangup Mixmont is not working ,Whenever my give code is executing i got following error: file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory app_mixmonitor.c:286 mixmonitor_thread: Cannot open /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140211/4d1aa4b0/attachment.html&g...

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would

We have a customer where their VM running Asterisk appears to have crashed. Fortunately, we had some debugging enabled. The asterisk messages file has this... (in notepad+ the blank line in the middle is all [NUL][NUL] [NUL][NUL]....) [08/12 15:30:55.880] VERBOSE[6920] app_mixmonitor.c: Begin MixMonitor Recording CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f [08/12 15:30:55.881] VERBOSE[6921] bridge_channel.c: Channel CBRec/IS__a37ae004-c780-4c7f-88a9-a04402f0ab4e-0000e70f joined 'softmix' base-bridge <23340bca-6823-4c70-a395-e3b092aeb671>...

Hi I'm using asterisk 1.8.3.2 (with a couple of patches) I have the following scenario... SIP call comes in and gets answered by extension A (MixMonitor is executed as part of this inbound dial plan of the number being called) Extension A puts call on hold and calls extension B Extension A then does an attended transfer of incoming call to extension B I'm finding that the recording

...able info available. #1 0x00000037e8071fac in free () from /lib64/libc.so.6 No symbol table info available. #2 0x000000000046b7b7 in ast_frame_free (fr=0x1b9da4b0, cache=0) at frame.c:369 No locals. #3 0x00002aaab1173573 in mixmonitor_thread (obj=0x1bb08220) from /usr/lib/asterisk/modules/app_mixmonitor.so next = (struct ast_frame *) 0x0 write = 1 mixmonitor = (struct mixmonitor *) 0x1bb08220 f = (struct ast_frame *) 0x1b9da4b0 fs = (struct ast_filestream *) 0x2aaac80f3b70 oflags = 577 ext = 0x1bb08466 "wav49" errflag = 0...

...MI. When I call MixMonitor I store the channel name in a var and then I use StopMixmonitor from AMI sending the stored channel name as parameter. What I've seen is that the app returns failure and going a little bit deeper I see that the failure comes from the function stop_mixmonitor_full in app_mixmonitor.c datastore = ast_channel_datastore_find(chan, &mixmonitor_ds_info, S_OR(args.mixmonid, NULL)); if (!datastore) { ... return -1 I know the error comes from that !datastore but I do not know how to follow and dig into the problem. any help? cheers, Jon -- PekePBX, the...

...risk/confbridge/278731.wav)") in new stack [Oct 27 15:08:48] VERBOSE[14698][C-0005278c] pbx.c: -- Executing [s at TTMConferenceTalker:8] ConfBridge("SIP/sbc1-0004e881", "278731,default_bridge,TTM_profile,TTM_profile_menu") in new stack [Oct 27 15:08:48] VERBOSE[14730] app_mixmonitor.c: == Begin MixMonitor Recording ConfBridgeRecorder/conf-278731-uid-1917613225 I see that we are setting it on the both sides of the conference with the same filename, but it has been working fine till asterisk 13.? -------------- next part -------------- An HTML attachment was scrubbed......

...BOSE[21378][C-000105c6] file.c: -- <SIP/243-00018dab> Playing 'custom/beep26_converted.slin' (language 'es') [Apr 16 13:42:42] VERBOSE[21378][C-000105c6] res_musiconhold.c: -- Stopped music on hold on SIP/5547740414-00018da4 [Apr 16 13:42:42] VERBOSE[21568][C-000105c6] app_mixmonitor.c: == Begin MixMonitor Recording SIP/5547740414-00018da4 [Apr 16 13:44:57] VERBOSE[21378][C-000105c6] pbx.c: -- Executing [h at from-internal:1] Macro("SIP/5547740414-00018da4", "hangupcall") in new stack [Apr 16 13:44:57] VERBOSE[21378][C-000105c6] pbx.c: -- Executing [...

...l the aforementioned delay had passed. "au" and "sln" had the lowest latency (3 seconds), so I'm using "au" for now. Is there any way to reduce the startup latency and make MixMonitor write the audio stream to the output file immediately? I looked briefly at apps/app_mixmonitor.c and main/file.c but I don't fully understand the code. Is mixmonitor forking an external conversion process to generate the audio data? thanks for any insights! -- G. Paul Ziemba FreeBSD unix: 9:06AM up 10 days, 11:05, 4 users, load averages: 1.39, 1.50, 1.54

...eout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28759 - A non negotiated rtp frame causes call...

...eout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28759 - A non negotiated rtp frame causes call...

...-- Executing [record at sub-record-check:5] Return("SIP/1002-00000292", "") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [out at sub-record-check:3] Return("SIP/1002-00000292", "") in new stack [2013-09-25 13:57:33] VERBOSE[9746] app_mixmonitor.c: == Begin MixMonitor Recording SIP/1002-00000292 [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [09999999999 at from-internal:5] Macro("SIP/1002-00000292", "dialout-trunk,1,9999999999,") in new stack [2013-09-25 13:57:33] VERBOSE[9745] pbx.c: -- Executing [s...

...eout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_suppor...

...eout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_suppor...

...eout is erroneously triggered during direct-media (native_rtp) bridge (Reported by Michael Neuhauser) * ASTERISK-20325 - Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples. (Reported by Olivier Krief) * ASTERISK-28780 - app_mixmonitor: Memory leak due to race condition between AMI MixMonitor and hangup (Reported by Joshua C. Colp) * ASTERISK-28773 - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active (Reported by Torrey Searle) * ASTERISK-28769 - DTLS Handshake Fails to Occur if ice_suppor...

...ion is available (Reported by Jeremy Kister) * ASTERISK-25622 - WARNING for "JABBER: socket read error" should be more specific (Reported by Sean Darcy) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rt...

...error" should be more specific (Reported by Sean Darcy) * ASTERISK-26515 - rtp_engine: Allocate RTP payloads on a per-session basis (Reported by Joshua Colp) * ASTERISK-26818 - cdr: Problem setting variables in h exten (Reported by scgm11) * ASTERISK-26875 - app_mixmonitor: Recording out of sync when 183 but no RTP (Reported by Aaron An) Improvements made in this release: ----------------------------------- * ASTERISK-26088 - Investigate heavy memory utilization by res_pjsip_pubsub (Reported by Richard Mudgett) * ASTERISK-26427 - res_hep_rt...