similar to: GUI client for windows for live monitoring/barge

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]

-----Original Message----- From: woody+asterisk@solutionsfirst.com.au [mailto:woody+asterisk@solutionsfirst.com.au] Sent: January 12, 2004 11:25 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] GUI client for windows for live monitoring/barge > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com]

In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my

I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started

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 =>

Hi All Can anybody help me for dial plan to barge or Spy(ExtenSpy) specificor selective extemsions among 20 extension in my office. lets say my office extension range is 301-320 & i want to barge only 3 extension say 320, 302,314. is this possible to barge specific extension? . Plz help me for this.I am using Asterisk 1.4.9 & SIP channels. Regards Amit -------------- next

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have a couple of questions on the zapbarge: 1) zapbarge asks for a channel - how would a manager know what channel to enter ? Is there any way of being able to enter an extension number instead ? I know that you can get the information from the manager interface, but I wouldn't want to give my users access to this, or have to install / write a system just to get an extension number from a

I'm running Asterisk 1.6 and was wondering if anybody have a workig "barge in" solution running. I was thinking of using chanspy, but i would like that the original call would be dropped, and the new call would be the only one there. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. 2. As an extension to the above, while on call, can the agent request a conference from another agent and later hang him up. 3. Is there any way for a call to be put in the queue destined for a specific agent

Hi there, I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine :-) But I am wondering if there is a solution/application which will enable me to implement voice recognition while playing a voice file (barge in). So that the caller hears a voice file and can interrupt it with his voice. Currently (on our platform) the caller has to wait for the end of the voicefie. Then we play

Hi, I'm planning on migrating someone who uses a very mature system. They would be logging in either as AgentLogin() or AQM. The main requirement however, is: The supervisor will have a control panel, where he will see how many of his agents are on call. If they are, he can "right-click" on the agent and get the options Call Monitor (where the super just listens in on the call,

Hellow, i need the following requirements with asterisk : 1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? 2) How call barging can set up in asterisk ? 3) How call recording can set up in asterisk? Thanks mahboob System Engr SSL -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All Can anybody help me for dial plan which can barge inbound call groupwise. Because when i am trying to barge inbound calls which is coming on my DID number i can hear 1st 3 digit of my Inbound provider IP address instaed of extension which pick that calls. I tried Chanspy as well as Extenspy. But result is same. So Plz Help me. Thanks Amit -------------- next part --------------

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Sorry that got accidentally sent incompleted, here's the full post: OK, here is the long drawn out description of how I am using Zap Barge and Monitor: Zapbarge(listen in on live calls): Very simple actually I just added this to my dial plan(extensions.conf): ; barge monitoring extension exten => 8159,1,ZapBarge exten => 8159,2,Hangup Then when you dial 8159 on

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

Hi All, I encountered a DTMF problem. We have an IVR built on Asterisk 1.0.7 with RedHat 9. When the caller calls into our IVR, and IVR plays the first prompt and asks caller to dial four-digit extension. Caller has to dial slowly, otherwise, Asterisk cannot recognize the extension number. I look at the trace on Asterisk CLI and there are missing digit in the middle of string. ex, caller

I know you can set up monitoring of queued calls, and I'm pretty sure my question's been answered before, but has anyone devised of a way to actually barge into a queue channel so you can do in place monitoring of calls? -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

Hi, is it and how is it possible to live monitor (barge - in) a SIP to SIP call without any Zap Interface? I am using asterisk 1.0.5 with chan_capi from Junghanns and SIP clients. I was looking for chan_spy application but it seems to be no longer available. Bye, Sven -------------- next part -------------- An HTML attachment was scrubbed... URL: