search for: stopmixmonitor

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......

...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 Doug Lytle <support at drdos.info> wrote: > > On 3/9/19 9:56 AM, Gokan Atmaca wrote: > > a) work for recording incoming / outgoing calls > > b) do not work for recording internal calls > > then we might be...

...e without called is not interesting. The uploading will fail since the MixMonitor is still on when I try to upload the file. The file will not have a duration. It works when I schedule the uploading a while after from my agi application but I would rather not rely on a timeout. When I tried to run StopMixMonitor before the Agi call in the h extension, the first call fail and I never get any uploading with callid. -- Executing [s at macro-ccdev2-rec:1] MixMonitor("SIP/upps-ccm-tq01-00000043", "cbrec-15607.wav,b") in new stack == Begin MixMonitor Recording SIP/upps-ccm-tq01-0000004...

Hi all Using asterisk 16.25 I was trying to stop Mixmonitor using features. The code is executed but I realized that I was executing StopMixmonitor from another channel so I opted to use AMI. When I call MixMonitor I store the channel name in a var and then I use StopMixmonitor from AMI sending the stored channel name as parameter. What I've seen is that the app returns failure and going a little bit deeper I see that the failure comes f...

...[applicationmap]] but it doesn't seem to work. Pressing #1 do nothing. Here my dialplan: 9999 => { Set(__DYNAMIC_FEATURE=pauseMonitor); MixMonitor(test); Dial(SIP/1000 at srv01,30,TtX); } [applicationmap] pauseMonitor => #1,self/both,stopMixMonitor Any advice? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140827/058a8edf/attachment.html>

...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 tried adding a StopMixMonitor after calling the macro but that did not help and I have tried putting an Answer() at the start of the dial plan. I am currently using the Monitor application instead which is working OK but I?d much rather stick with MixMonitor. Asterisk version is now 11.17.1 Thanks! From: Mark Farmer Reply-...

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

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 weekend, none of them seem to get the CDRs right. Ideally I'd like the...

Hi all, I need to use asterisk to record all phonecall I have test using mixmonitor to record a call. Now I need to set the configure file to let asterisk auto record all calls. I have searched many document but still can not succeed. My version is 1.8beta and I prefer using mixmonitor. Regards!

...n,Playback(custom/queue-sales-no-operators) same =>n,Hangup() same =>n(checksurvey),GotoIf($["${MMSURVEY}" = "1"]?survey,s,1) same =>n,Hangup() exten =>h,1,NoOp(When a sales queue call is hung up) ; note CDR is already written by this point same =>n,StopMixMonitor() same =>n,Hangup() [survey] exten =>s,1,NoOp(Take the survey) ; note CDR is already written by this point same =>n(q1),Read(MMSURVEYQ1,custom/survey-q1,1,,,5) ... more survey here -- Mitch

...orted 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 release: * --- Fix Sorting Order For Parking Lots Stored In Static Realtime (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel (Closes issue ASTERISK-21294. Reported by daroz) * --- When a session timer expires during a T.38 call, re-invite with correct SDP (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) * --- Fix white noise on SRTP decrypti...

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 =>

Hi everybody, I've been working on a project which records the voice of the incoming call. I use record_file function of asterisk as described below: RECORD FILE <filename> <format> <escape digits> <timeout> [offset samples] [BEEP] [s=<silence>] filename: record1 format: wav escape digits: # timeout: -1 offset samples: 0 BEEP: 1 silence: 3000 *Please read the

---------- Forwarded message ---------- From: Arun Kumar Chaudhary <uniquearun04 at gmail.com> Date: Sat, Jun 21, 2008 at 4:51 PM Subject: Detection of Answer, hangup,busy etc while using Dial command To: asterisk-users at lists.digium.com. Hi Guys, I am in kanpur, India. I am using Dial() command in my phpagi script. I am unable to detect whether it is connected to the dialed number, if

> Show us the parts of your dial plan which: > > a) work for recording incoming / outgoing calls > > b) do not work for recording internal calls > > then we might be able to give you a clue what's wrong. Hello For example: My phone number is 1000, the other's number is 1001. These numbers are in the same PBX (asterisk). I want 1000, 1001 can call to record audio

Is anyone can help me with this ? I'm really desperate. Thx in ad. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ikka - Mitra Kreasindo Sent: Wednesday, September 14, 2011 5:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Mixmonitor command parameter problem on

How do I make a user dial a passcode if he wants to make an international call? ________________________________ This electronic message contains information from BOSH Global Services which may be company sensitive, proprietary, privileged or otherwise protected from disclosure. The information is intended to be used solely by the recipient(s) named above. If you are not an intended recipient,

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to