similar to: (possibly) new use for asterisk

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

From: "Nik Martin" <nmartin@radiancetech.com>> To: <asterisk-users@lists.digium.com>> Subject: RE: [Asterisk-Users] asterisk compile problem Date: Wed, 14 Jul 2004 09:22:38 -0500 Organization: Radiance Technologies, Inc. Reply-To: asterisk-users@lists.digium.com Fletcher Bonds wrote: >> Hello all >> >> As of 5pm PST today (7/13), I pulled

Of course, big brother wants his say in the matter. http://www.wired.com/news/politics/0,1283,63884,00.html?tw=wn_2polihead

Can the Manager Redirect command transfer a ringing SIP extension? I'm trying to implement a Camp On feature, and having failed to do it in Dial Plan logic, am trying to do it with manager logic. If an arbitrary Sip extension is ringing, I need the ability to pick up that extension from any other phone. What little docs there are on Manager commands shows Redirect takes these parameters:

Hi all, I get these errors while trying to run RealProducer 8.5 in WINE: err:heap:HEAP_ValidateInUseArena Heap 40310000: in-use arena 4036ae80 next block has PREV_FREE flag err:ntdll:RtlpWaitForCriticalSection Critical section 0x40310070 wait timed out, retrying (60 sec) fs=0297 err:ntdll:RtlpWaitForCriticalSection Critical section 0x40310070 wait timed out, retrying (60 sec) fs=0297

> Well, I thought about it as well but up to now didn't try. Via which > way do you intend to have e.g. a line-in to the iPaq with reasonable > quality. Would be really interesting to figure out such things ... > and then use a WLAN-cradle to transport data to the internet *g* The main use would be covering events at sports fields, auditoriums, etc. that are away from the

I'm having a tough time getting call pickup to work on *. Here's my configuration: X100P with T-1, channels 1-4 voice <---> * <---CISCO 7960 with SIP 6.0 Image A call comes in, and * picks up and presents a menu. Caller chooses extension, (in this case ext 103, SIP/wsmith) Wsmith is sitting in my office, hears his phone ringing, picks up my phone, gets dial tone, and presses

Dudes.... I'm running Debian sarge and i can't get the %$#%# sound to work. I installed Starcraft but it doesn't recognize any sound at all. My sound card is a SB extigy. HELP PLZ! Ignacio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://www.winehq.org/pipermail/wine-users/attachments/20051208/bf4d5ccf/attachment.htm

Hi All, Couple easy questions: 1. How do I view what environment my app is running under (i.e production / testing / development) - I''m not sure what RAILS_ENV is set at, how to I take a look at it? 2. Somehow (?) I have turned logging "off". I''d like to have it back on :s I have files called production.log, test.log, server.log, development.log in /myapp/log/ but

I''m using Restful Authentication, and the code to create a user is pretty straight forward - there is a before_save action and a before_create action: before_save :encrypt_password before_create :make_activation_code But for some reason when I try to create a user programmatically in the controller like this: User.new(:email =>

I’m using a Windows development component which uses vorbis.dll, ogg.dll, vorbisenc.dll for encoding an Ogg Vorbis file. It's all working well except for one user occasionally has a 1 hour file appear as 2 hours and it plays at half speed. It is being converted from stereo to mono before feeding the encoder with a channels=1 configuration. Here is an example file which will be available for

In the absence of "The Definitive Guide to Asterisk Dial Plans" book, I'd like to do something possibly unique with the formatting of extensions in my dial plan, and am having trouble. We use VoicePulse connect, which gives us local DID for inbound and outbound calls (even though DTMF tones are not working in Voice Pulse Connect at the moment). To dial local numbers, you have to

Hi I've got the error "cannot allocate a new connection -- maximum of 16 connections already opened" after I tried to create a new connection to a database. However, the reason ist, that i did not disconnect previous connections.... I don't know the name of this connections. How can I disconnect this "unknown" connections and drivers? if I delete all objects, the

Hi all, i have a problem: i have an adsl modem that is connected to internet. I can''t manage this modem. Between my PC and the modem i have a linux firewall that make the NAT and the traffic shapping. I have create a script that limit the bandwidth of the "external" interface of the firewall so i can manage my bandwidth for my internet application. The problem is that i need to

I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about "ast_get_txt" not found. Recompiling and

Hi Guys, Other than than yum repository (which fails when installing freepbx with it) are there any automated install scripts out there that would install Asterisk 1.6 or 1.4 onto a CentOS LAMP system? If the script install FreePBX that would be a BONUS. Thanks, Bruce -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings, Does anyone know of a tool that can act as a VoIP client and stream to a streaming server such as shoutcast/icecast, etc. I've got a client interested in doing basketball play by plays during tourney season. They have * in place now and the bandwidth to burn for streaming out. In the old world, I did an analog phone patch -> mixer -> encoder -> streaming server. What

Hi Nik, Flushing MDB can only be done when we are managing it per station not per port. For that we need to have MCAST_TO_UCAST, EHT and FAST_LEAVE. Here one more point is - some vendors may offload MCAST_TO_UCAST conversion in their own FW to reduce CPU. So, the best way is to have MCAST_TO_UCAST enabled and MDB will become per station, so we can delete MDB on disconnect. Shall, I create one

Yes, I downloaded the aumix utility and am using it now. Set line in to %50 then %25 and to record but still the same thing. I'm streaming from the line in on the sound card which is being fed from a portable radio nearby. I get the distortion whether I have the audio fed into 'line in' or not. Matt -----Original Message----- From: owner-icecast@xiph.org

If I have an IAX client (Firefly softphone in this example), and the client is not registered at the moment because they are not connected to the network and someone dial that extension, they get the user's "I'm on the phone at the moment" message vs. the "I'm unavailable" message. Is this by design? Here's the extension in question's dialplan: