similar to: DTMF issue on IVR

Hi all, I got into a strange problem here. I've got an asterisk box with bristuff-0.2.0-RC7k, and a HFC PCI ISDN card, running in NT mode. The ISDN card is connected to a S0 bus and to a Siemens ISDN PBX. Two phones are connected to the ISDN PBX and are successfully getting calls from the asterisk box. When dialling from one of the phones, the ZAP channel seems to be missing out on some of

hello, I am planning a fairly large hotel VoIP system, using analog phones. It will consist of about 100 analog phones, that must have access to a VoIP server. I am considering an option to use a couple of asterisk boxes, bundled with a total of four TDM2460E cards, and one TDM2451E card. Has anyone on this list done something similar? It would be great to hear some comments regarding a smilar

Hi there, I am using redhat 9.0 with asterisk 1.0.7. I created an user and was be able to leave voice messages to that user and retrieve the voice message. I looked the wiki and setup the voice message as the email attachment. However, I have never received email with the voice attachment. Here is the setting for voicemail.conf: ; ; Voicemail Configuration ; [general] ; Default formats

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:

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

-----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]

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

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've seen a few but can't get them to work. I need one where I can drop a call into a conference without them knowing it to us it as a live monitor and barge function, anyone doing this are know of a gui client for windows I can use? Thanks,

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,

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:

I noticed when I call certain IVR systems, such as 1800calldhl, that Asterisk will not barge the prompt. Would this imply that Asterisk has an Early media detection problem. Is anyone else experiencing this problem. Is there a fix? Kurt

Greetings all, I've tried out chanspy, but what I'm really looking for is the ability to interrupt a call (i.e. barge in for emergency purposes). Has anyone found a way to do that with Asterisk? Regards, Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060403/dfe6e99e/attachment.htm

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

Everyone we have a few new things to give back to the asterisk community. http://bugs.digium.com/bug_view_page.php?bug_id=0002379 http://bugs.digium.com/bug_view_page.php?bug_id=0002380 http://bugs.digium.com/bug_view_page.php?bug_id=0002381 These include app_chanspy, the ability to spy on ANY bridged call taking place inside asterisk. NOT just ZAP as with ZapScan/Barge. Native format_* files

If SLA supports IP trunks, can shared extension appearance be achieved using a local SIP trunk in place of an extension? Basically, I'm trying to allow some stations (Polycom IP 650) to have a shared extension amongst all of them. Ideally, I'd like for the LED to show if that extension is in use, and I'd like for the extension to ring all stations on that extension when a call comes

Hi, This is a naive question, but is there a way in my AGI script to simultaneously play audio and listen for DTMF or voice responses? I've heard VOIP hackers call this "inbargeability;" it's the ability to "barge in" to a playing audio clip. I'm planning to use Lumenvox for the DTMF and voice recognition, BTW. Not sure if that matters. Many thanks to anyone who

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: