similar to: ChanSpy by anthm and more...

ClueCon is coming in 2 weeks so we urge everyone who plans on attending to register today so we get a proper headcount! ClueCon was put together by Asterlink, the same team of people who helped shape Asterisk into what it is today by writing features, fixing bugs, offering IRC support and assisting with the management of the development effort. We have produced several real-world solutions based

Hi, Anyone know anything about asterlink? I've heard good things about them. So I thought I'd give them a try. I went to sign-up.... however, I haven't heard from them, I've called them several times between 9am and 5pm EST and no responce. Called today again after 9am EST, no responce. Tried Sales, Billing, and CS, and got the same person's voicemail! Is this a oneman

I have a situation where I have a number of fields and a rating for the field. So, in a DRY way, this works fine: <%= [''abc'', ''def'', ''klm''].each do |e| %> <tr><td><%= @result[e] %></td> <td> <%= @result[e+''rating''] %></td> </tr> <% end %> Now, I

Hello I have an old Asterisk where I need to listen to Agent calls. So I created this code: exten => _555,1,ChanSpy(Agent) exten => _555,n,Hangup() But I always get: 2009-10-22 16:00:38 WARNING[5695]: pbx.c:1720 pbx_extension_helper: No application 'ChanSpy' for extension (default, 555, 1) It seems that Asterisk doesn't have ChanSpy enabled... is this possible? Which

Hi folks, We have a very rare problem with Asterisk 1.2 where Chanspy reports the following: Oct 31 19:53:29 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2015-11b21148 Oct 31 19:53:42 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2066-11d42b80 Oct 31 19:53:46 NOTICE[10490] app_chanspy.c: Attaching SIP/105-48807010 to SIP/2022-11c10170 Oct 31 19:53:58

asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap/4|q) exten=> 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304 at flash:1] ChanSpy("Zap/31-1",

Hello list, I was looking for a way to implement non-blind call transfers with *, i.e. the usual behaviour of most PBXs when pressing the flash button: - A and B are talking - A pushes flash - A is free to compose a new number - B hears music on hold - A's call is answered by C - A hangs up - B and C are in conversation As much as I can understand, * only supports blind transfers, where if

Hi All, I have installed ChanSpy but its not compiling gives me the following error....if anyone can help that will be really appreciated... [root@Asterisk asterisk-1.0.9]# contrib/scripts/astxs -install -autoload apps/app_chanspy.c make: *** No rule to make target `apps_env'. Stop. -c apps/app_chanspy.c -o apps/app_chanspy.o Thanks Umair.

I have read about that on asterisk 1.6, there will be a parameter "o" (Only listen to audio coming from this channel), I have tried, but I still get inbound and outbound audio from the spied channel. Has anyone used this feature? Is it working? Is there any work-around? I will like to only spy the outbound audio from a channel, I dont want to hear the incomming audio of that channel. I

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

With the deafening silence from my previous questions, I feel seriously alone in the desire to have ChanSpy available. I want to be able to perform a "ZapBarge" on an Agents conversation, and ChanSpy was the answer to my prayers. Bug #2379 (http://bugs.digium.com/bug_view_page.php?bug_id=0002379) was closed "bkw918 10-27-04 17:06 Closed pending new changes in cvs-head."

Oh -- then I did not understand the documentation. fallback-mount This optional value specifies a mountpoint that clients are automatically moved to if the source shuts down or is not streaming at the time a listener connects. Only one can be listed in each mount and should refer to another mountpoint on the same server that is streaming in the _same streaming format_. I was thinking that

I am trying to construct a macro for long distance dialling. I have two internet feeds, I have all routes including Teliax on Internet A and a static route to Voxee on Internet B. I thought I could use the dialplan entry below which uses the ChanIsAvail() command to check the connection, but this returns the provider but not the username, so I don't understand how to use this for real

hi, exten =>6000,1,dial(SIP/6000,15,tr) exten =>6002,1,dial(SIP/6002,15,tr) exten =>6004,1,dial(SIP/6004,15,tr) exten =>6006,1,dial(SIP/6006,15,tr) exten =>6008,1,chanspy(SIP/6006 | wbq) when i dial 6008 ,it is connected ,but i can't able to hear the voice of the any one. when coversation between the 6002 to 6006. in my Console mode i got the following comment *CLI>

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Hi guys, I'm writing an application that will sniff audio streams going across the network and allow a seperate client application to listen into one of these streams. I was looking into ways to distribute the captured stream media to a client for listening, and icecast came to mind. Icecast, as I understand it, is usually used for streaming a few data sources to MANY clients. As I want

Hi list, I'm having some strange problems since I updated to CVS HEAD three hours ago... First: I was using Iax Comm in some PCs, it suddenly stopped working, what I get is som pieces of audio once in a while, I mean instead of listening to the ring tone and then the voice on the other side I just hear a bit of the ring tone, maybe another bit, a bit of someone answering... like

Our company is at the point now where we're almost ready to switch over to an Asterisk server for a number of telephony applications. There is one final application I've been trying hard to find to replace something we already use with another provider. It's kind of an advanced "FollowMe" application. (freedomvoice.com) It works as follows: 1. An outside caller calls into

I've attached a small patch against icecast 2 which converts ice- headers to icy- headers for clients that include icy- headers in their request. This allows a few clients (notably xmms) to pick up stream info they otherwise miss. -b -------------- next part -------------- Index: src/format.c =================================================================== RCS file:

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches