similar to: Recording just first part of call?

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 =>

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.

Is there an equivalent stopmonitor command if you are using MixMonitor ? StopMonitor does not seem to have an effect on MixMonitor Julian.

I'm in the process of specifying the hardware for some new Asterisk systems which will be running a substantial number of conferences with recording. I was wondering what there is to choose between SCSI, SAS and SATA disks, in terms of performance for this kind of application. I will be using dual drives with kernel-based software RAID1. Any advice from experience would be appreciated!

In article <20151125133008.6369360.14455.17239 at gmail.com>, Israel Gottlieb <isrlgb at gmail.com> wrote: > Try putting progress instead of answer Yes, I tried Progress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to

Thanks for the suggestion Tony, I installed each codec for MoH, core sounds, and extra sound packages. Unfortunately the tests produce the same results. [Sep 1 20:36:45] ERROR[10081][C-00007fe5]: frame.c:343 ast_frdup: FRACK!, Failed assertion Excessive refcount 100000 reached on ao2 object 0x20380b0 ( continuously for a while followed by a [Sep 1 20:36:46] WARNING[7761][C-0000770d]:

In article <E02FA554-9D6D-4E7D-8A78-5FBDE1DE939D at kicp.uchicago.edu>, Valeri Galtsev <galtsev at kicp.uchicago.edu> wrote: > > > > On Sep 26, 2020, at 8:05 AM, Jerry Geis <jerry.geis at gmail.com> wrote: > > > > I have a disk that is flagging errors, attempting to rescue the data. > > > > I tried dd first - if gets about 117G of 320G disk

Does anyone know how a program, script or shell user can best determine whether the machine is running on bare metal or is a VMware guest? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org

I have been writing a lot of AGI programs in C with good success. I would like somehow to have an AGI program continue in the background while the pbx execution returns to the dialplan and continues. Is this possible? I was thinking that perhaps I could fork or create another thread within the AGI prog. The reason I want to do so is in order to monitor external information (e.g. credit limit and

In article <301a2e78-d490-3805-e30f-41b668aac5c1 at sysnux.pf>, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > > Hi Tony, > > Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > > Are there any other languages/libraries I should be considering? > > Same here, after years of AGI / AMI, I recently made my first project > using ARI on Asterisk-16. I love

I've just set up a new box with FC1+updates and the latest Stable Asterisk from CVS. Asterisk is started with the default safe_asterisk script with a console on TTY9. The coloured text on this console is made up of weird characters instead of normal. Please see http://www.softins.co.uk/dsc00018.jpg for an example. If I do "asterisk -rvvvvv" on a normal login, either via the

In article <55D20981.7030902 at centos.org>, Johnny Hughes <johnny at centos.org> wrote: > On 08/17/2015 10:57 AM, Tony Mountifield wrote: > > I recently applied updates to a CentOS 5 box running MySQL. I've discovered > > that the new version of openssl, 0.9.8e-36.0.1.el5_11, breaks MySQL SSL > > connections. > > > > If I rename

Has anyone else experienced a problem with the LED for span 1 on a TE410P or TE405P? I had a TE410P on which the span 1 LED would not light red, but once the span was connected, it did correctly light green. I RMAed the board to our UK distrbutor and received a replacement. However, the replacement board displayed the same problem! Wondering if it was related to the computer I was putting it

I want to use fail2ban on CentOS 6 to monitor Apache with the standard default logfile format ("combined"). Has anyone here succeeded in doing so? The format has the IP at the start of the line, followed by two dashes (if no authentication) and THEN the timestamp. What I've read on the fail2ban wiki seems to say that the timestamp must ALWAYS be at the start of the line, followed by

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

Has anyone here used Asterisk inside a WatchGuard firewall, talking via the WatchGuard SIP Application Layer Gateway to an outside SIP service? I have a customer doing just that, and I am 100% convinced there is a bug in the ALG regarding the media port number it inserts into the SDP when it rewrites it. However, either they or WatchGuard will not accept there is a bug, despite my very detailed

Hi, despite some searching I haven't found an answer to this question: Is there a way I can see in the log, or find out in the dialplan, what codec has been negotiated for a SIP channel? If possible, I'd like to do this in both Asterisk 11 and in an old 1.2 system. What I'm specifically trying to do is to determine historically the usage of the G.729 licences installed in a system,

In article <dafd6c5f-9676-c4da-8761-619d6f003525 at microlinux.fr>, Nicolas Kovacs <info at microlinux.fr> wrote: > Le 01/10/2019 ?? 09:05, Liam O'Toole a ??crit??: > > [...] > > > > FWIW, the current release has squidguard 1.6 > > https://packages.debian.org/source/stable/squidguard > > And this all begs the question: who's flying this plane?

Hi, I want to take a SRPM that is available for various versions of Fedora and rebuild it on a CentOS 4 system. Which release of Fedora is the closest to CentOS 4? In other words, which would be the best FC to take the SRPM for? Cheers Tony -- Tony Mountifield Work: tony at softins.co.uk - http://www.softins.co.uk Play: tony at mountifield.org - http://tony.mountifield.org