similar to: Macros not working as expected with extension "h" in some circumstances

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

I thought I would post this in case it might be of any use to anyone. Not anything special but it does work. Keep in mind you need sox and wmix. Here is some relevant exerpts of my extensions.conf using John Todds macro. [globals] CALLFILENAME=foo FOO=foo CALLERIDNUM=foo [default] exten => 287,1,Macro(dial,SIP/agent20002|20) exten => 287,2,Voicemail(u287) exten =>

On this macro I keep getting exited non-zero on s,3, but s,3 is doing what it is suppose to do but the macro stops. Is there a way to make a macro ignore errors and continue to s,4? I have the lattes ver of sox 12.17.4. Also if I just run this line from the command line I don't get an error. [root@redhat monitor]# sox in.wav in-rev.wav reverse [root@redhat monitor]# [macro-record-cleanup]

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian

Hi, i want to use soxmix to record some calls in my PBX. If i use soxmix on my linux shell it works so i can mixed two calls into one consolidated call. I want to do the process automatically since extensions.conf but it doesnt work. My extensions.conf looks like this: exten => 407,1,SetVar(MONITORDIR=/var/spool/asterisk/monitor) exten => 407,2,Monitor(wav,${TIMESTAMP}.${CALLERIDNUM}.wav)

Hi Folks, I've been playing with call recording for our support department which was kinda going ok until I spotted something odd in the CDR. None of the support calls are being entered into the CDR properly. I'm using mysql as the back end and Areski's web based front end and all was going fine. The problem seems to be that the CDR doesn't get populated with the destination

On Jul 29 2017, Guus Sliepen <guus-NnCthlHDAqpg9hUCZPvPmw at public.gmane.org> wrote: > On Sat, Jul 29, 2017 at 12:47:02PM +0200, Nikolaus Rath wrote: > >> >> Proxy = exec /etc/tinc/rath/triv_proxy.sh >> > >> > Indeed, it seems like Proxy = exec was broken by a commit that fixed >> > some issue with the other proxy types. For now the workaround is

On Jul 11 2017, Guus Sliepen <guus-NnCthlHDAqpg9hUCZPvPmw at public.gmane.org> wrote: > On Mon, Jul 03, 2017 at 02:24:55PM +0200, Nikolaus Rath wrote: > >> After upgrading my client system from Debian jessie to Debian stretch >> (which includes an update from tinc 1.0.24 to tinc 1.0.31), I am >> having trouble with my VPN: > [...] >> Proxy = exec

Hello, I am investigating a data-loss bug in SSHFS (https://github.com/libfuse/sshfs/issues/72). While the root cause is in SSHFS and has already been fixed, there seems to be some unfortunate interaction with what I believe is an OpenSSH bug: As far as I can tell, when sending a SSH_FXP_OPEN request with SSH_FXF_WRITE (i.e., opening the file write only), and then following up with a

Hello, After upgrading my client system from Debian jessie to Debian stretch (which includes an update from tinc 1.0.24 to tinc 1.0.31), I am having trouble with my VPN: As long as I let tinc connect directly (no "Proxy" configuration option on the client), everything works fine: # tincd -n rath -D -d tincd 1.0.31 starting, debug level 1 /dev/net/tun is a Linux tun/tap device (tun

The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting

Currently at our office, if I want someone else to pick up a call, I have to transfer the call to them. So I'm looking into call parking, which is ALMOST perfect. The missing piece of the puzzle: I'm extension 203. I want any call I park to get parked at extension 2203. I want a call my boss parks to park at 2205, since he's ext. 205. In other words, I want calls parked FROM

Hi all, What's the best method to send automon call recordings (*1) to the voicemail box of the Asterisk user? Do you have to trap hangups, etc, or is there some global variable that can be set? Thanks! S.

Hi All, I'm working from home today (DSL -> Internet -> 2MB leased line -> A*K server behind NAT), and trying to pickup voicemail using Zoiper.. I can access the VM system, I hear all the prompts, and I can even hear part of the message playback. But then I get silence on the call (call stays up), and I get: Parsing

agibson@nova.org wrote: > > I just brought up samba 2.0.5a in a Solaris 2.6 environment. Smooth > effort with no visible problems in CVS download, configuration, and > make but attempts to reference data after connection yield this > error. Has anyone seen this before and can point me in the right > direction to to address? Thanks Andy > > [1999/07/31 15:14:30, 1]

hello friends, I am configurin my Billion ISDN and when I start asterisk (asterisk -vvvc) I have this error message: [chan_zap.so] => (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line: No '=' (equal sign) in line 29 of zapata.conf Apr 23 15:27:23 WARNING[2205]: config.c:525 process_text_line:

Using DIAX softphone which seems to be working OK can get to VM/echotest etc in the demo context Am trying to setup FWD but get the following problems Can hear it ringing when dialing FWD no 612 for time. Connects but no sound from remote end. Does anyone have any suggestions. Softphone on 192.168.0.2 asterisk on 192.168.0.3 Netgear RP114 doing NAT to the internet port 5060 being forwarded to

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

I seem to be having a conceptual problem with the "While" and "EndWhile" applications. It seems that on the first cycle, even if the result of the "While" is false that the enclosed applications will get run. Is this expected? It seems to be counter-intuitive, but I don't know what the intent of the While routines is. I could of course put a

Hi all, I need to (quickly!grrr) update IP addresses for which a given machine will answer. I did this: /sbin/ifconfig eth0:0 xx.xx.xx.10 netmask 255.255.255.0 /sbin/ifconfig eth0:1 xx.xx.xx.11 netmask 255.255.255.0 /sbin/ifconfig eth0:2 xx.xx.xx.30 netmask 255.255.255.0 /sbin/ifconfig eth0:3 xx.xx.xx.31 netmask 255.255.255.0 /sbin/ifconfig eth0:4 xx.xx.xx.13 netmask 255.255.255.0