similar to: Asterisk 1.6.2.0-beta4 - Monitor / MixMonitor Recording

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 folks, I was using the following featuremap: blindxfer => *1 disconnect => *9 atxfer => *2 parkcall => *7 automixmon => *0 and everything worked. Then the need arouse to use some features like automixmon during a conference, but MeetMet has the * key bound to the (admin) menu. Thus, in order to enable features like automon and transfers even during a conference, I

Hi all, I have 2 quick question regarding the file location and post record command of the recording using automixmon in features.conf. With the normal monitor/mixmonitor applications you can change the location of where the recordings will be stored, by changing the MONITOR_FILENAME variable. I tried changing the TOUCH_MIXMONITOR_OUTPUT variable to include a path but it sill puts the recorded

Hello, I have been attempting to get the Monitor function to accept a loal variable substitution in order to use the same filename later in the same context. Monitor does not appear to like it, as it attempts to use wav|filename as the recording type, as opposed to just wav. Here is what I get if I just supply a filename directly (it works fine): --context----------------------------- exten

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Hi list! I'm trying to configure Asterisk to record incoming calls, if the called press *3. I added in features.conf: automixmon => *3 then, in my dialplan: exten => 1,n,Dial(SIP/00493511111111,20,RcxX) Well, if I **CALL** a number I'm able to record the call, but if I'll be called, and press *3 nothing happens... In the console I can't see anything, too. Could you

On 1 July 2016 at 17:41, Joshua Colp <jcolp at digium.com> wrote: > > >> exten => 1234,Set(CALLERID(all)="Jon Doe" <+123456789>) >> same => n,Dial(PJSIP/phone123, 30) >> > > Your exten line has no priority, is that how it is in your dialplan? > Actually no, I stole that line from an earlier email to this list. Mine has a priority.

hello liste i have creat i trunk sip and inboun route for inbound calls the issue whe i use the DID in inboud route i have a error No DID or CID Match. but when i leave this DID field blank i can route the call without any issue how can ido in order to use DID in route inboud "i use elastix" Executing [s at from-trunk:1] NoOp("SIP/358-106-000000c0", "No DID or CID

The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues

The Asterisk Development Team is pleased to announce the third beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta3 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This is an incremental release of the 1.6.2.0 branch as the previous beta was released just over a month ago, and many issues have been resolved since then. Included in this release are the following issues

The Asterisk Development Team is pleased to announce the second beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been

The Asterisk Development Team is pleased to announce the second beta of Asterisk 1.6.2.0. Asterisk-1.6.2.0-beta2 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release merges in changes to the device state code which caused a performance regression in Asterisk 1.6.1 and 1.6.2. The result of this device state code review is that performance has been

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

The Asterisk Development Team has announced the release of Asterisk 1.6.2.0, and Asterisk-Addons 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.0 is the first feature release since Asterisk 1.6.1.0, which was released April 27, 2009. Many new features have been included in this release. For a

The Asterisk.org development team is pleased to announced the release of Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release 1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons 1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of Asterisk-Addons is compatible with both Asterisk 1.6.1 and 1.6.2 branches. These releases are available for

The Asterisk Development Team is pleased to announce the first beta of Asterisk-Addons 1.6.2.0. Asterisk-Addons 1.6.2.0-beta1 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This beta fixes a several issues with chan_mobile from the chan_mobile refactor branch, and issues related to cdr_addon_mysql and cdr_config_mysql. Additionally, this beta is compatible with

we tested asterisk 1.6.2.0, found that when call from one sip_channel to another sip_channel , ------------------------------------------------------------------ exten => _X.,1,Noop() exten => _X.,n,Dial(SIP/${EXTEN},50,TtgM) ------------------------------------------------------------------ in asterisk 1.6.2.0 ,when sip user config to use dtmfmode=rfc2833 , it's ok, but when both

hi,list Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after i make and make install. i cant find the .so file. is this mean it can't install on 64bit Cent-OS. ps: it works fine on the 32 bit Cent-OS Thanks very much! -- Thanks for your supporting, have a nice day. Sucan

Hi all, We encountered a strange phenomenon when trying to upgrade from 1.6.2.0 to any newer releases: We use the following cli command to feed a wave/mp3 file into an existing conference on an other serve: /opt/asterisk/sbin/asterisk -r -x "channel originate Local/ConfGongAdmin at XY_Features extension ConfGongPlay at XY_Features" The corresponding extensions.conf part looks like

The Asterisk Development Team is pleased to announce the the second release candidate of 1.6.0.11, the release of 1.6.1.2, the first release candidate of 1.6.1.3, and the fourth beta of 1.6.2.0. These releases are available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ . The release of 1.6.1.2 fixes a remote crash security vulnerability in the RTP stack. The