similar to: Asterisk mixmonitor with 16khz

(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed,

(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed,

Hello, I try to find informations concerning Mixmonitor command, but ... without success. MixMonitor command take at last parameter "command". This command can be a shell script. When record is over, and this command executed, asterisk wait for a return code or asterisk move to the next dialplan instruction ? This command is a background task or use ressources in asterisk ? For

I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in server. I am having some issue in audio quality. I want to enable jitter buffer on asterisk but don't know, how to do. Any one can help me. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130423/015bd0de/attachment.htm>

Jean-Marc Valin wrote: > Hi, > > I just released unstable version 1.1.2 that contains more fixed-point > work. Though it's still not 100% complete, enough have been done to make > it run in real-time on ARM. In order to do that, compile with > --enable-fixed-point --enable-arm-asm. All narrowband modes work in > real-time with complexity 1 (some work with higher

Hi , you need to install these packages(epel repo enabled): yum install attr bind-utils docbook-style-xsl gcc gdb krb5-workstation \ libsemanage-python libxslt perl perl-ExtUtils-MakeMaker lmdb-devel libarchive-devel \ perl-Parse-Yapp perl-Test-Base pkgconfig policycoreutils-python pygpgme \ python2-crypto gnutls-devel gpgme-devel jansson-devel libattr-devel keyutils-libs-devel \ libacl-devel

Jean: The 6711 is actually a floating point processor so it can do the floating point without emulation. I am working on the 6416 which does not have it. Working with fixed point may still speed things up possibly. Muhammad: It seems like you might have to start getting into linear assembly to speed things up. If you are serious to getting speex working on TI DSP we should work together on

Hi all, Can asterisk 1.8.x give me MixMonitor recordings of 16Khz sampling rate? Any help would be appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120801/ee8ac7a6/attachment.htm>

Hi Rowland It's the same error in rhel 7.6 I have tried that Also I compile without gpgme and now I can provision but still the domain are not there!! From: Emad Yousuf Said Al Kharusi Sent: Sunday, March 31, 2019 5:14 PM To: Rowland Penny <rpenny at samba.org>; samba at lists.samba.org Subject: Re: [Samba] error in samba 4.10.0 while using samba-tool domain provision

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, 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, We are recording our calls to queues by putting the appropriate options in our "queue.conf". This is all working properly. We would now like to set the MixMonitor option to adjust the caller volume (which is very quiet). With the regular MixMonitor application, we would just add the "v4" option to make it much louder. I don't see a way to set this option when

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

Hi, sorry to bump this one but I still have this problem. The file is always created but is always zero size. This is the dial plan that records the call: exten = _0[1-8]X.,1,Set(CALLFILENAME=/var/spool/asterisk/callrecordings/its/${STRFTIME(${EPOCH},,%Y/%m/%d)}/Outbound-${UNIQUEID}) exten = _0[1-8]X.,2,MixMonitor(${CALLFILENAME}.gsm,b) The dial plan then calls a macro that makes the call. I?ve

Hi, I am facing some issues while using MixMonitor and StopMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(num)}_${TIMESTAMP}.gsm,b) exten => s,2,Dial(SIP/101,13) exten => s,3,StopMonitor() exten => s,4,NoOp(Dial Status: ${DIALSTATUS}) exten => s,5,Goto(sss-${DIALSTATUS},1) exten => sss-NOANSWER,1,VoiceMail(777 at salesvoice) exten =>

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to

Hi everyone, I'm trying to get calls to record with the following setup: Using phpagi originate is called from a web application: $asm->originate("Local/" . $row['extension'] . "@sip-standard", $row['phone_number'], "sip-standard", "1", "", "", "7000"); The agent being called is extension Local/101 at

Greetings. Just updated from 1.4.22 to 1.8. Minor changes in dialplan and things work ok. Except for one thing. I have a call to MixMonitor. This is implementing a dictaphone kind of app. With forwarding recordings to email and storing them on the server. The process works so that we dial into Asterisk and answer the phone, initiate MixMontior and WaitExten until recording finishes. Problem is

Hello, I recently switched over to using Mixmonitor versus Monitor to see if it would clear up some warble that I was getting in my recordings. It did indeed clear that up, but a new problem was introduced. The recordings for no reason will just end abnormally. There is no rhyme or reason as to when they will end, but usually after a minute or so. Here is my current setup. Asterisk v. 1-2-0rc2