similar to: dialplan / problem with extension-length > 1

hi everybody, I'm currently planning a small-sized web-applicaiton allowing users to call-in via phone. the phonecalls should be recorded and processed further by some custom scripts - sounds like asterisk is a perfect match for this app. however, during prototyping I have no ISDN-connection whatsoever available, so I was asking myself if it's possible to connect a cellphone via

hi list, I'm looking for a way to execute commands in my dialplan specifically when a caller has hung up. my curretn dialplan looks like this: exten => s,1,Answer exten => s,n(restart),BackGround(intro) exten => s,n,Read(Enter,4,4) exten => s,n,Voicemail(${Enter},u) exten => s,n,agi(process.php|${Enter}) exten => #,1,Playback(thanks) exten => #,n,Hangup it lets a user

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

I ran into a problem today while trying to compile chan_cellphone version 17 on a CentOS 4.4 machine. Apparently the bluez and autoconf versions were to old and as I tried to install the latest version, I found that the new bluez-lib would install and allow the chan_cellphone to compile, but bluez-utils required an update to D-sub which in turn required python 2.4 or better. That apparently in not

Hi, I download the last svn and I also look around but I cannot find the source, I only found the patch http://bugs.digium.com/print_bug_page.php?bug_id=8919 any one can help me out. thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm

28th February I am working with Asterisk 1.2.15. I have configured sip.conf for two soft phones (I am using Xlite).I have installed the Bluez stack and so far, i manage to make a phone call from a soft phone to a GSM network. However, i have an audio problem. The soft phone can be heart by the GSM costumer but the voice in Xlite is not transmitted to the GSM. In asterisk all i got is the

Hi, What is the actual functionality of ezstream when I use it with Icecast?What does it exactly do? Why isn't Icecast written to read a file directly and stream it? -- ~$ubh

On Mon, 2005-06-27 at 19:33, Michael Kamleitner wrote: > hi, > > icecast 2.2.0 has to be patched slightly to work with flash-players using > the sound-object - cant find the exact link to this, maybe one of the > developers can explain? I just added 4 lines at the end of > format_mp3_send_headers() in format_mp3.c: > > sock_write(client->con->sock,

Hello, I've created a macromedia flash player in order to play icecast mp3 streaming. Icecast is configued on the port 8000 and It's ok with winamp. In my flash movie, I've used loadSound("<streaming>",true) on a Sound object. It's ok on Mozilla based browsers but some user don't have sound on Internet Explorer. D't understand? Is there an issue? Thx

Hi all I noticed that most caller are quite confused by the standard voicemail announcement text. Especialy as the number read is the 'internal' number. Callers often hang up because they think having called the wrong number when they hear the announcement. Is there a way (like in many other PBXes) that the VoiceMail user could record his own announcement? (like, hello, this is the

Hello! I have been googling a lot and asked wiki a few times now, but i cant find a howto for setting up a voicebox. Any link/hint would be great! Thanks, Mario

hi everyone, we're running a webradio-station focussed on electronic music (http://www.play.fm), using icecast 2.2.0 (on SuSe Linux 9). we're broadcasting 4 hours of live-program each day (using streamtranscoder). what I would like to do is to stream a generated playlist of mp3-files during the rest of t. day (using something like ezstream, which I have currently installed & played

hi folks, I've built a simple PHP-script utilizing the AGI-interface. in extensions.conf I trigger the script and pass a single value as first argument: exten => h,1,DeadAGI(process.php|${Enter}) On the Asterisk-console, I can actually see that the script is called correctly (something like "DeadAGI(process.php|1234)"). However, when I read stdin in the PHP script, I receive

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by

I'm running chan_cellphone version 13 on the latest svn trunk (as root). I believe I have chan_cellphone set up correctly (bt addr and port retrieved from the "cell search" CLI command). When I load the chan_cellphone module, my Motorola V3m asks if I want to allow "Asterisk PBX", I say yes and enter the 0000 for the pin, then my phone tells me the pin is invalid. Here

Did not know how to make up a subject line for this. I have a dial plan that allows a caller can try my cell phone. And that's fine. If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail. Is that possible? joe a.

I am having difficulty with the calculation of the power law exponent for set of nodes within a graph. Specifically, I am interested in the distribution of in-degree and out-degree among communities of web pages where the web pages are the nodes of the graph and the hyperlinks the edges. According to the literature, the distribution of incoming and outgoing links obeys a power law distribution

I'm currently looking to interconnect my Asterisk PBX system with the PSTN via a digital PRI/T1. I know a multitude of options exist for internal PCI cards (Digium/Sangoma/Rhino), I was wondering if anyone has any experience or recommendations of external PRI media gateways that support SIP. So far I've found: VegaStream Vega 400 Audiocodes Mediant 2000 MediaTrix 1531 However they are

Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier --------- MBDSYS SARL 1, centre commercial de la Tour