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...the book but still can't figured out what is missing, so please give me some suggestion. Thanks for all input and your time. Best Regards, Nguyen __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com

sorry if this has been asked before. is it possible to configure asterisk to use an outbound sip proxy? thanks __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

...e dial statement to sip), i can answer the incoming call without a problem. is this a known bug? (asterisk cvs 05-03-04 using RedHat v9 on Via mini-ITX) __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

(1) What is the "standard" MIME tag for FLAC files? I found "audio/flac" and "application/flac" in various postings in the newsgroups. (2) What MIME tag (if any) does the Winamp plugin support when dropping URL's? I'm trying to serve up a link to a FLAC file and drop the link on the Winamp playlist. This works fine for MP3 files but I can't seem to get

I'm new to Asterisk and have been attempting various configurations. I'm having problems with the basics of SIP to SIP phone communications within my own network. I've configured two phones ( Xten X-Lite) and whenever I dial either one I get errors as follows: *auto-congestion SIP/Phone 1 *SIP/Phone 1 is circuit-busy *Everyone is busy at this time I usually don't post such

...you want to retry?" I know that I have the no password mistype, can you help what's when wrong? Thanks for your time. Best Regards, Nguyen __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Hi, I got a samba 3.0.2-6.3E (normal rpm) server running on Redhat Enterprise 3.0. since yesterday I can't connect to the printers share to administrate the printer drivers. When I try to connect I get the following error in the log file: [2004/05/06 10:36:26, 0] lib/util_sock.c:read_socket_data(342) read_socket_data: recv failure for 4. Error = Connection reset by peer [2004/05/06

...ere a way to automatically install printers to windows clients when users log in? Maybe by running a logon script... Thank you in advance. __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

...Is there a timeout that I can configure in smb.conf file or such to reduce the delay ? Any suggestions/pointers, please email me. Thank you very much. __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

...ec 2 what am i missing? i promise i will write/publish a complete "* with 2 x x100p's and softphones for very very dummies" guide!!! tia -k __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Hi, I am new in asterisk and i've bought a X100p and a TDM400... First of all, how can i verify my config files ? Secondly, when i'm trying to pass a call to the outside, i ve a Notice about appdial.c (l 554) telling me: unable to create channel of type Zap ...and i don't understand... Finally, when i plug my analog phones in RJ45 of my TDM400, there is no tonality ( i'm not

...er = 206.67.210.5 hosts allow = 127.0.0.1,206.67.210.,192.168.53.,10.222.7.,10.222.8. case sensitive = Yes map archive = No Tony __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

...rds somewhere? I've got ldap sync set to on but that doesn't appear to be working. Any help would be greatly appreciated Kevin kbmetz@yahoo.com __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Hi all; I have been searching for an answer to a question that a customer asked me and I have only found a few older answers. So, wanting to find out if anyone has any experience with this issue and can help provide me with some advice. I have a customer which is strongly interested in using Asterisk as a PBX. One of the core requirements, however, is that the system MUST be able to

...ssion, nchannels 2 debug1: channel_free: channel 1: x11, nchannels 1 debug1: Calling cleanup 0x80628b0(0x0) ================================ Thanks, Ryan __________________________________ Do you Yahoo!? Win a $20,000 Career Makeover at Yahoo! HotJobs http://hotjobs.sweepstakes.yahoo.com/careermakeover

Hi, I have just upgraded my Cisco 7960 phone to SIP firmware today and I have to say it's working great with Asterisk. At work (which uses Cisco Call Manager), when a voicemail is recieved the read light remains lit until the voicemail is retrieved. Is there any way to achieve the same effect with Asterisk ? Thanks, Paul. -------------- next part -------------- An HTML attachment was