similar to: Can't stop Mixmonitor

Jon, We are only using FastAgi. On the second system (running Asterisk 16) there are no agi's running (just some bash scripts on call hangup). I did add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep -v udp6 | grep -v ' 0 0.0.0.0' | grep udp) to my bash script to check out the packet queue (with the help of

Anyone have any other ideas? On Tue, Oct 27, 2020 at 1:27 PM Dovid Bender <dovid at telecurve.com> wrote: > Jon, > > We are only using FastAgi. On the second system (running Asterisk 16) > there are no agi's running (just some bash scripts on call hangup). I did > add some hackey code (netstat -nua | grep -v 'udp 0 0' | grep > -v udp6 | grep -v

El Wed, 8 Jul 2020 12:23:42 -0700 John Kiniston <johnkiniston at gmail.com> escribió: > Dovid, You could use func_odb + a ODBC Redis driver to keep from having to > shell out. > > DO you know odbc redis drivers? It would be nice to store cdrs ans other stuff in redis without patching asterisk -- PekePBX, the multitenant PBX solution https://pekepbx.com

El Tue, 9 Jun 2020 09:46:32 -0600 George Joseph <gjoseph at digium.com> escribió: Hi George > > > > > > If transcoding is enabled Would it be possible to do the same but handle a > > 488 > > back from Bob and failover to another INVITE with Bob's allow list to > > handle > > transcoding? That way we would always try no-transcoding before

El Tue, 27 Oct 2020 12:52:47 -0400 Dovid Bender <dovid at telecurve.com> escribió: > Hi, > > Sorry in advance that I am emailing the users list and not the biz list I > think I will find my target audience here. We are looking to hire a > consultant to help us figure out an issue. We are having what seems are > "random load" issues with bare metal boxes that are

Hi all Using asterisk 16.25.0 (I know it's a bit old) I'm trying to parse the realtime queue_log and I realized that not every call has a ENTERQUEUE event. I see that if there's an internal call to an extension that has a dialplan forward to a queue (no call is placed to the extension before calling the queue), the ENTERQUEUE event is not generated in the queue_log For calls from

Hi, Sorry in advance that I am emailing the users list and not the biz list I think I will find my target audience here. We are looking to hire a consultant to help us figure out an issue. We are having what seems are "random load" issues with bare metal boxes that are dedicated to Asterisk and a few Perl AGI's. We went after all the usual suspects (CPU IDLE, memory usage etc.). I

Hello list, how does StopMixMonitor know which 'monitoring channel' to stop when there are multiple conversations that are being monitored/recorded ?? I want to use StopMixMonitor in a macro, called from within applicationmap (features.conf). Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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})

I went from asterisk 1.4.22 (which was working) to SVN and I am getting the message rejected because extension not found... How can I modify the print statement in chan_sip.c line 18388 to include not just the extension but the context its trying to find my extension in??? ast_log(LOG_NOTICE, "Call from '%s' to extension"

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, 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

Hello Mynum: 6001 , Othernum: 6002. I can record as follows. But I do not enter individual records for each internal required. I want to do it more smoothly with a Macro. Thanks. exten => _6001,1,NoOp() exten => _6001,n,MixMonitor(${UNIQUEID}.wav,ab) exten => _6001,n,Dial(SIP/6001,20) exten => _6001,n,StopMixMonitor() exten => _6001,n,Hangup() On Sat, Mar 9, 2019 at 6:50 PM

Hello *, is the rtpip patch still valid for asterisk 1.6 (with some code changes, obviously)? https://issues.asterisk.org/view.php?id=8161 Or, in asterisk 1.6 there is an alternative to using it? This is the difffile I produced for chan_sip.c in asterisk 1.6.2.11 --- chan_sip.c 2010-10-12 13:47:49.000000000 +0200 +++ chan_sip.c.orig 2010-10-12 13:47:27.000000000 +0200 @@ -987,9 +987,6 @@

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

asterisk 11.1 Documentation in cdr.conf for endbeforehexten reads: Normally, CDR's are not closed out until after all extensions are finished executing. By enabling this option, the CDR will be ended before executing the "h" extension and hangup handlers so that CDR values such as "end" and "billsec" may be retrieved inside of of this extension. I have

The Asterisk Development Team has announced the release of Asterisk 11.4.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.4.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

I''m testing how xen and grub2 can work together. I''m using xen 3.4.2 and grub2 1.97+20091115-1 from debian sid. After many failed attempts I''ve discovered that on my system the first argument that grub2 should pass to the linux kernel is lost. This is my working grub2 configuration: menuentry "Xen 3.4.0 / Debian GNU/Linux, Linux 2.6.31-1-xen-amd64" {

I''m testing how xen and grub2 can work together. I''m using xen 3.4.2 and grub2 1.97+20091115-1 from debian sid. After many failed attempts I''ve discovered that on my system the first argument that grub2 should pass to the linux kernel is lost. This is my working grub2 configuration: menuentry "Xen 3.4.0 / Debian GNU/Linux, Linux 2.6.31-1-xen-amd64" {