similar to: G729 experiences..

Does G.729 provide better voice quality than GSM? (a question for people who have tried both) --J. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 188 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030715/52f18205/attachment.pgp

Hi, I'm looking for a good codec to use on a personal VoIP setup. It is strictly for my personal use, I'll never resell it, make money or it, or whatever. It seems a free personal-use G729 codec is available as a WIN32 library. I find it puzzling that at the same time one has to pay license fees to use it under Linux, even non-commercially. I was wondering -- am I missing something?

Hi I thought I share this one, just in case this is an indication of some bug ... When I was trying to use music on hold at first, I didn't bother to copy any music into /var/lib/asterisk/mohmp3 since there was a sample- hold.mp3 in there which played just fine in a standalone MP3 player. But after uncommenting one of the lines in musiconhold.conf and doing reload on the console, there

Hi, Do the g729 codec licenses for Asterisk work on a SIP environment (only SIP UAs running g729 + Asterisk)? I would like to buy a couple for a SIP test lab but I have not found any documentation on wether it works for SIP UAs or not. The Digium page only mentions: "The G.729 codec works with all Digium cards." Can somebody tell me please? Thanks, Ricardo Villa

Hi, I am thinking about getting some of these phones to use with my Asterisk system.. So I would be interested to hear from the people who have tired and tested these phones.. What is the voice quality like using alaw/ulaw or G.729?? Are there any problems getting them to play nicely with Asterisk?? How sturdy is the construction (reciever, handset, buttons, cable connections, etc..) ?? How

Hi, Does the G.729 module support adding more licences??

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Cisco should really be ashamed of this product... While it is physically well constructed, and has excellent sound quality along with a very pleasant user interface, the device has SERIOUS stability issues, unless you run your network with an iron fist... Quite by accident, while configuring my Asterisk system to connect to a Cisco 7960 via SIP in a standard office PBX type arrangement, I

Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of asterisk@[IP Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by

Hi All! First, let me introduce myself, as this is my first post to the list (I've been lurking for quite some time now). My name is Matt Hardeman, and I work for a software development firm in Birmingham, AL. We are interested in the Asterisk PBX and it's various configurations first as an internal solution for our occasionally bizarre telephony needs, and eventually are interested

Has anyone written a script that can be used as a cron job or similar that will test if Asterisk is running and if not restart it?? I have just had an issue where asterisk crashed and someone was trying to call me.. it would be nice if it could have been automatically restarted.. I was thinking of a simple bash script something like running "ps -aux |grep asterisk" and then some kind

Hi all: I've just purchase some licences of G.729 codecs, and I like to bring up * using /etc/rc.d/init.d script. Does anyone knows how to start in the "old" way? Thanks in advance, Gus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031016/6dd07c4b/attachment.htm

Hi, I have just got my SNOM 200 to start doing some real testing with *.. I am trying to use the GSM codec but the quality is really bad, Is that normal? does anyone actually use GSM?? Also are there any 'gotcha's' that I need to look out for so I don't spend hours trying to get somthing working that really doesn't work anyway.. Thanks.. later.. --

Hi.. How do you accept user input while waiting or playing moh? My Dialplan is as follows.. ring,ring,.. Hello thanks for calling blah blah... Please enter the extention number blah blah... WaitMusicOnHold(10) If no input pass call to operator.. The problem is that the user has to input the extension while they are being told what to do.. any input during Wait or WaitMusicOnHold is ignored...

I see in the changelog that ENUM support has been added.. anyone know what this is? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Is Asterisk (or can it be set up as) a SIP proxy? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a "basic" Linux/Asterisk system? Just re-boot and config. -- James Taylor jltaylor@metrotel.net 903-793-1953 --

Look like SNOM have rolled back the firmware version of the 200's from 1.16w to 1.16q.. Anyone know why? -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

Hi, What is the correct chan_capi dial syntax?? This is what I think it is.. exten => _90.,1,Dial(CAPI/[msn here]:${EXTEN:1}) This seems to work for local numbers.. but I have an access number for cheap long distance calls.. wich gets dialed and then the number I want to call is sent as DTMF after a few waits (w).. On my X100P I used the following.. exten => _9001.,1,Dial(Zap/1/[access