similar to: let incoming callers contact a certain extension...

How to play GSM files? I want to go through some of them but I'm not sure which player to use. -- #Joseph

Hi. I have installed a x100p (THE x100p for those who have seen my former post). Now I just want to connect a "normal" phone (not an IP phone) to the card and use it as a sip extension (I have a FWD account)... more clearly: I want to be able to pick up the phone and call any FWD user using my FWD account... receive the FWD calls in that phone, and also to be able to make normal

Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include => demo And demo is

Hi. Can you give me some hints on how I can create a rotational board? I dont even know how to spell it in english. What I want is to have more than one line reserved, but with a single phone number, so that people can call to the same number and get a ringing signal if any of the lines is available, instead of having to dial 5 different numbers in order to get a free line. This is done

Hi list Can someone tell me what the exact CentOS policy for end-of-life is for 5.0? Let me explain: I have been using RHEL for a number of years and had to switch to Fedora for installing some packages only supported under F6. As I find Fedora's policy for release life duration too stringent (and I do no longer need the packages that made me choose F6), I'm considering switching over

Can anyone supply a link on a how-to (or provide intructions) on recompiling a centos 4 kernel? I'm trying my old rhel 3 commands and they arent working.

I am trying to set up an auto attendant for the first time, and am having trouble getting to the submenu. My extensions.conf file looks like this: [incoming] exten=> s,1,Background,menu1 exten=> s,2,Wait,20 exten=> s,3,Goto,s|1 exten=> 1,1,Playback,option1 exten=> 2,1,Playback,option2 exten=> 3,1,Playback,option3 It is my understanding that asterisk treats the digits entered

Hi, Is there any further information on this UPS interface? I have an Upsonic UPS that is reporting the same interface, but I cannot make it work or get any sensible information from it. Thanks Jon --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000706-2, 25/01/2007 Tested on: 26/01/2007 11:00:14 AM avast! is copyright (c) 2000-2007 ALWIL Software.

Hi, Is there any further information on this UPS interface? I have an Upsonic UPS that is reporting the same interface, but I cannot make it work or get any sensible information from it. Thanks Jon --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 000706-2, 25/01/2007 Tested on: 26/01/2007 11:00:14 AM avast! is copyright (c) 2000-2007 ALWIL Software.

Hi Alexander, thanks for your good work on the megatec_usb driver. I have made a few more (cosmetic) changes. Is the driver now working stably for your device? In particular, do the instant commands (and particularly "upsdrvctl -k") work properly? In this case, we could hopefully mark the device as "supported" and merge the driver to the main branch. There is still the

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Hi. I'm new to Asterisk and have been working on setting up a development server but have gotten myself a bit confused. I'd like to implement the following logic for calls coming from the PSTN: Check for caller-id yes => keep going no => play SIT and prompt for telephone number Check time of day to see if it's day / night day => ring some phones

Hi Jon, your patches are malformed because (presumably) your email client wrapped this lines. Could you please send these as a unified diff ("diff -u"), and send them in attachments, rather than as a copy-and-paste? Thanks, -- Peter Jon Gough wrote: > > Peter, > Here are the files I have worked on. I hope I have diff'd them > correctly. I have not touched the

Peter, I have been doing some more testing and I find that if I modify the libusb.c file to set a report size in libusb_get_report to 16 the error messages go away. This is the code as it currently stands: static int libusb_get_report(usb_dev_handle *udev, int ReportId, unsigned char *raw_buf, int ReportSize ) { int ReportSize_test = 16; unsigned char raw_buf_test[ReportSize_test +

Hi all. I've being reading posts from the list since yesterday and I feel this question was answered a lot time ago, but the list archives are a mess (yet). I hope some one is willing to help me out. I want to set up this: caller ----- PSTN ---- (SOMETHING1) ------ VoIP --------- (SOMETHING2) ---- PSTN I think this must be a very basic architecture, but I'm not sure wat SOMETHING1

Hi everyone, I haven't used Asterisk in many years, but in searching for a good podcasting solution that will allow me to record three or four participants to individual tracks (which would allow me to go in and do noise removal on each participant individually, giving a higher quality), I came up with the idea to use Asterisk. Now I've installed it and got it all set up and did a test

Hello, As a podcaster I use Asterisk extensively and often have several people in a conference room. We'll record the calls via a SIP phone connected to a sound mixer. Is there an easy way to bump up the audio bitrate for all callers connected to the Asterisk server and improve the general sound quality? The server is not used much outside of recording the podcast. We're not opposed to

Does anyone know of an efficient (or at least polynomial in the worst case) way to map a local DS node to all complete and/or global nodes it may correspond to in any of the function's callers? Thanks, --Patrick -- If I'm not here, I've gone out to find myself. If I get back before I return, please keep me here.

Most of the time, when someone calls in from the outside on a POTS line, and possibly over IAX as well, they don't hear any ringing sound while the internal SIP phones ring. If you call from an inside SIP phone, even forcing it into the incoming context, you hear the ringing. The outside calls can answer and talk fine; just no ring indication. Is there a setting that controls this?

is there something that needs to be set up to make the 'called' and 'callers' buttons work on this phone? all i get is the backlight to switch on and off. Jason Kawakami