similar to: asterisk 13 mixmonitor - random missing syllables

Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> hi, >> >> i have strange problem with asterisk 13 mixmonitor, recording to wav >> (centos6) >> when the system is under load, there are sometimes missing syllable >> >> there arent BIG spikes on cpus >> recordings are to ramdisk (/dev/shm) >>

Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to wav >>>> (centos6) >>>> when the system is

Dne 28.1.2016 v 13:37 Brian :: napsal(a): > when you say load - how many concurrent calls? Is there transcoding > happening? sip / PRIs ? what load? > 12 concurrent calls no transcoding SIP under 1.5 with 4x 1Ghz vcpus (its vmware VPS) > On Thu, Jan 28, 2016 at 9:57 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > >

I tried to Build LLVM under Syllable, but it was failed on the next moment http://clip2net.com/clip/m0/1246547164-clip-99kb.png any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20090702/346df366/attachment.html>

hi, which file extensios are supported in mixmonitor application? https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Application_MixMonitor can i record to Opus? -- --------------------------------------- Marek Cervenka =======================================

hi, i want convert mixmonitor recorded speech audio from wav to mp3 or aac can you recommend your settings for speech audio? filters, noise elimination, compression ratio, ... i will probably use lame thank you -- --------------------------------------- Marek Cervenka =======================================

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

doesnt work for me Dne 29.5.2016 v 17:48 Niklas Larsson napsal(a): > Hi, > > > On 2016-05-27 18:28, Marek ?ervenka wrote: >> after downgrade to 13.8.2 >> May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip >> b49162cd sp bfac0940 error 4 in >> libmysqlclient.so.16.0.0[b48f1000+12e000] >> >> after downgrade to 13.7.2 >>

Dne 13.8.2015 v 17:20 Rusty Newton napsal(a): > On Thu, Aug 13, 2015 at 3:54 AM, Marek ?ervenka <cervajs at fpf.slu.cz > <mailto:cervajs at fpf.slu.cz>> wrote: > > hello, > > is it possible simultaneously use chan_sip and chan_pjsip? > > if yes, can you recommend settings > > i'm thinking about > - chan_sip - for sip

hi, i have the same problems as in https://issues.asterisk.org/jira/browse/ASTERISK-25833 my current combination is centos 6 32-bit, unixODBC 2.3.2 (recompiled from fedora20), mysql 5.1.73, mysql-connector-odbc 5.1.5, asterisk 13.9.1 i tried update to mysql-connector-odbc 5.3.6 from oracle but it segfaults every time can you share your working mysql+odbc combination? thanks odbcinst.ini

hi, after switch from 13.7 + pjproject 2.4.5 to 13.9.1 pjproject bundled i have problem with segfault (centos 6) Program terminated with signal 11, Segmentation fault. #0 0xb7665695 in check_cached_response (sess=0xafbd688c, packet=0xb07676d8, pkt_size=132, options=1, token=0xafecc2bc, parsed_len=0x0, src_addr=0xb0e47a20, src_addr_len=16) at ../src/pjnath/stun_session.c:1287 1287

after downgrade to 13.8.2 May 27 18:21:06 ast kernel: asterisk[16286]: segfault at 1010024 ip b49162cd sp bfac0940 error 4 in libmysqlclient.so.16.0.0[b48f1000+12e000] after downgrade to 13.7.2 asterisk is ok Dne 27.5.2016 v 18:09 Marek ?ervenka napsal(a): > btw info from my segfault > > Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. > Program terminated with signal 11,

Dne 4.2.2016 v 12:17 A J Stiles napsal(a): > On Thursday 04 Feb 2016, Marek ?ervenka wrote: >> hi, >> >> is there way to get SQL schema for Asterisk 13.7.0 without alembic? >> thanks > Assuming you already have Asterisk up and running, you can just use > > $ mysqldump -d -uroot DATABASE TABLE1 TABLE2 TABLE3 ... > > will print (on STDOUT, so you can just

hi, i'm evaluating performance of centos7 i did tests on centos6 x86_64/distro kernel 2.6.32, asterisk 11.16.0 with 500calls (7sec alaw, simple dialplan, pass trough - sipp generators/asterisk receiver with answer/playback) scenario - sipp generators - asterisk - asterisk receiver (i wrote ansible scenario for this if you are interested) then i reinstalled system to centos7 x86_64/distro

Hi, I have a simple Queue(named 1) and one Member(SIP/1119) logged into it. Now when a caller is placed into Queue and gets connected with Member, I want to record the call. It does record the call when I use MixMonitor() before placing the caller into Queue, but not when MixMonitor() is used in macro which is called upon Member answering the call. Following is my dialplan... [mixmonitortest]

Hey, I've come across two interesting problems today. First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help. Now it seems that

Hi all, I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on

hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

Hello Asterisk list, Hope you are all doing well! We are using the MixMonitor application to record the calls and under some situations the call can be spied using ChanSpy with whisper enabled. Sometimes the spying channel is a person who can interact in the call, and some other times it is a sound file playing a message. The problem is that for some reason the MixMonitor does not record whatever

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.