similar to: voip

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

All I apologize if this is slightly off topic, but I'm in the process of trying to get NUT working and I've discovered the need for a serial port probe that is native db9 - native db9 (the thought of hanging a probe plus two db9-db25 adapters plus a cable on a UPS bothers me greatly). I'm not looking for anything fancy, just a little gender changer with a buncha LEDs on it.

In order to ease my students into the R environment I am developing a package which installs a variety of utility functions as well as slightly modified versions of some standard R functions -- e.g. mean, hist, barplot, .... In my versions of these standard R functions I either add options or alter some defaults that seem to create difficulties for most of my students -- for example, when they do

I'm attempting to refactor an R script that does a lot of plotting, among other things. Ideally I'd like to do something like setup # does pdf(...) for each part of input { plot(process(part)) } cleanup # does dev.off() but have problems: 1 I'm plotting to PDF, so everytime I dev.off() creates a new file, and I want everything in one file (as does my boss :-) 2 I'm

On Sun, 24 Jan 2016, Mark LaPierre wrote: > I'm forced to use MS Windows 7 at work. They have rolled in so much > smart phone/tablet stuff that it makes the desktop even more of a pain > to use than Windows XP was. Examples include, you can have your > applications any color you want as long as it's gray, and you can no > longer search for files by anything other than the

Hi, I''m looking at the rails cookbook for uploading files[1]. At the bottom of the page it says that the sanitize_filename functioin would probably go in a helper or a library. I wonder where these files would be or what they would be called. Pehaps there are some conventions I need to know about? Thanks, Peter [1] http://manuals.rubyonrails.com/read/chapter/78

Are there any sip providers out there providing full business telephone service. Not just single line/residential service like I have seen with vonage etc. For example take a company currently using a legacy pbx connected to the PSTN with a PRI. I would like to replace this setup with a data T1, an asterisk box, and some SIP Phones then pass all calls (local and long distance) directly

I sent this once already but it didnt show up in my mailbox so I am not sure if ever made it to the list. If it did, my apologies. I have an * system and an nbx. My plan is to use several grandstream 286s and plug the the phone cords into the NBX's analog FXO card. It works fine but after I make a call, the port stays open on the NBX and I cannot make another call for around five minutes

Hi? 1. When a mobile user dial an IVR short code , mobile network able to divert that call to Asterisk platform.? 2. There would be web servers which are holds Voice XML . 3. Asterisk would be able to redirect the mobile request to certain Voice XML server accordingly. Just for like this setup , how do we install asterisk with minimum of asterisk modules ? Do we need to install complete

Hi If I do not use any DAHDI - hardware , can I ignore the DAHDI linux and tools installation ? Can I use just asterisk ? is there any dependencies while executing Asterisk with DAHDI modules ? Thank you Luke -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130607/b7c8e748/attachment.htm>

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files . What are the configurations has to be made with asterisk ? Thanx in advance, Luke. Send instant messages

I am running Asterisk CVS-HEAD-07/14/04-16:28:29 and noticed that when I send a subscribe I get back a 403. This used to work in an old version which I forgot to record before upgrading to the above version. Any suggestion? I can register fine with the * server. Sip read: SUBSCRIBE sip:2486@192.168.0.2:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK46F2668 From:

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi folks: Does the TDM400 card from Digium only support FXS, or is FXO functionality available or planned? -- Ron Gage - Saginaw, Michigan I am looking for work - resume at http://www.rongage.org/resume.doc Electrical Engineering, Linux Programming, Networking

Anyone tested this card in Greece? I would like to buy several of these but I don't know if it works well with the PSTN in Greece. The Quicknet Linejack card seems to be unable to understand the telephone network's tones (busy, ringing,...). TIA, Michael.

Hello all, A newer version asterisk-oh323 (v0.5.5) is available. This version contains updates for compilation with latest Asterisk sources, some additions ("atexit" cleanup) and some other fixes. http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

Hi all, i don't know how often someone ask for this, but i ask agian: Is it possible to use G723.1 with * or not ? I tried to use G723.1 from * over OH323 to a gatekeeper from my provider. The situation is following: Zap/analog ---> IAX -----INTERNET-----IAX--->OH323---->GATEKEEPER/PROVIDER The provider supports G723.1. Can someone help me ? Regards, Thomas.

Hi, I am trying to set up * and ISDN-PRI (channels 1 - 15) using E100 boards. I installed zaptel and libpri. When I execute modprobe -r wct1xxp I get an error message: ZT_CHANCONFIG failed on channel 1: Function not implemented (38) /lib/modules/2.4.21-SAX/misc/wct1xxp.o: post-install wct1xxp failed /lib/modules/2.4.21-SAX/misc/wct1xxp.o: insmod wct1xxp failed Follows my /etc/zaptel.conf:

If you have an FXS port and would like to attach a PSTN analog line to it this device would do the job by converting the FXS port to FXO. It's a small external device. Works well with VOIP FXS and other FXS interfaces. Interface: 2 RJ11 Jacks (one for the FXS port and one for the PSTN outlet.) Cost: $35.00 with USPS regular mail included. Power: 9 to 20V dc power supply (Not