similar to: Bugs that Need Your Input!

Hello list partners you know about a softphone made in java attachable in a web page? GNU! Thaks in advance!______________________________ Visita http://www.tutopia.com y comienza a navegar m?s r?pido en Internet. Tutopia es Internet para todos. -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I read variables in AGI scripts, I see only the follwing 13 variables agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode beside these, I found following variables documented on several sites. agi_calleridname agi_callingpres agi_callingani2 agi_callington agi_callingtns Where can I

I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ? [02:14] <bkw__> Let me tell you how chidlish digium and Mark Spencer is. I walk into a restaurant with them all here at Astricon wearing my sangoma shirt and he asked me to leave. [02:15] <Dovid> u serious ? [02:15] *** mog

I am Jerry Richmond CEO of ByVolution LLC. We have purched some did's from you that we use to test with, weare going to order our first batch of 250 this week. John Blackman is not with us any more. I need for some one to call me on my cell phone because our office no. 9198270713 to 9198270720 can not except outside calls. Brian Sponaugle tells me he is not getting the help he needs from

Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks

hi we need a predictive dialer which can be used with asterisk software. Is it possible? Bye Owais Bin Zuber --------------------------------- Do you Yahoo!? Yahoo! Search - Find what you’re looking for faster. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040310/a49a4b5c/attachment.htm

Is anybody using 1.2 (or 1.2.1) in a production network using Realtime (voicemail, sip or extensions) with 100+ SIP phones? If so, what are your experiences? We've been running 1.0.3 for about a year and it's been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm afraid of killing our stability. Obviously, we'd do it in stages (upgrade to 1.2, then realtime

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

Thanks everyone for the input. After pricing everything we need out, it's not worth trying to get our old system to work, so I've pitched ditching everything and starting over. I'm very excited and hoping they'll go for it. Regardless, I'm going to throw a box together for my house, we have no home phone (just cell phones) so this'll be a great way of testing. All

Hello, How would I go ahead a record all phone calls into and out of my asterisk server. I know the legality issues behind it, but I could always play a recording to let people know they will be recorded. Brian J. Schrock Network Engineer, RHCE, CCNA Anistone Technologies Phone: 614-537-2817 FAX: 614-573-7165 6926 Avery Rd. Dublin, OH 43017

On 20/04/15 17:45, john wrote: > > > > Is this wheezy ? if so, it might be an idea to use backports, this > will get you 4.1.17 which is still in development, 3.6 is now EOL > OK, I understand a bit better where your problems lie. I would still use backports, supported code is (hopefully) better code :-) > > > I'd be willing to do that if it got me support

I've googled, RTFMed, and I still can't fix this. Hoping someone here can offer a suggestion. I'm learning, so please explain a little if you can. I installed samba on a Ubuntu 5.10 machine (named mog). My XP machine "sees" it, but the username/password dialog fails. I've traced it down to what I think is an authentication problem: mat@mog:/etc/samba$ smbclient -L

so becky said mog is out, i remeber seeing it about a year ago on craigslist but didnt apply since it sounded like a last.fm ripoff and why waste time reinventing a wheel?..anyways, im wondering if it performs ok for anyone? it keeps freezing up on my trying to edit the page. sometimes a response comes back after about 10 seconds. im only guessing it runs rails, based on the gratuitous use of

Hello all, I just updated my development Asterisk box today and am noticing weird behaviour in the CLI. Whenever I hit "tab" the CLI shows me the list of commands, but the output is garbled. See below. I'm running the v1.0_stable code, checked out about 2 hours ago. asterisk*CLI> show version Asterisk CVS-03/30/04-14:34:01 built by root@asterisk.n2net.net on a i686 running

Hello, On Saturday, Oct 2nd 2004, I did a 45 minute presentation on Asterisk to the Ohio Linuxfest. Roughly 350 people attended, and I think I spoke with nearly all of them at some time during the day. N2Net had an Asterisk PBX running on a Laptop w/ a pair of SNOM 200 Phones setup for people to play with. When all was said and done, nearly 800 incoming/outgoing calls were made through the

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

Hello, I have samba 2.0.7 installed on AIX4.3.3. I have 2 bugs : - the first is that I can't transfer large file (more than 4GB) : I've seen there is a bug and patch - the second one is a secure pb referenced by a colleague : a root access on the destination machine Are these 2 problems corrected on version 2.2.8a ? Thanks for your help. Sophie VERDAVAINNE NIS SchlumbergerSema * :

Hi. I would like to partially apply a function to a list of arguments, and I don't know how to do this in R, without perhaps writing default values to the formals() of my function, or writing to the environment object of a function. For context, my definition of partially apply is: "fix some of the arguments, leaving the others as variables, return a new function that takes the un-fixed

The phpagi site and package itself contains some examples that are really helpful scoot over to http://phpagi.sourceforge.net/ and grab it kr, Bart -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Cristian Draghici Sent: dinsdag 24 januari 2006 16:04 To: Asterisk Users Mailing List - Non-Commercial Discussion

Hello, I've got a mixture of SPIP 300 and 500 phones in production for various clients. I've got the XML settings configured for local conferencing, but I'm not seeing the expected behavior from the phone when I attempt to conference two calls together. According to the manual, while talking to the first party, you simply hit Conference, dial the second party and then Conference