similar to: Using Swat - Could not connect to host localhost (port 901) error!

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

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.

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

i tried using music on hold option in the dial command exten => 7777,1,Dial(zap/1/7777,20,m) when someone calls me and i picked up the phone, the call will be suddenly dropped. however, if i use a sip client instead of zap (also changing the 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)

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

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

(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

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

Hi Everyone, I'm runnging Mandrake 10, trying to learng samba. I have install Samba 3.0.2a sucessful with all the /etc/inetd.conf, /etc/services, and /usr/local/samba/lib/smb.conf files configured. I also restart smb, test the "smbclient %U -L localhost", and have root user add in samba using "smbpasswd -u root password". But when I tried to login using Swat as a root

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

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

Anybody knows what do I need to tell Asterisk to issue a re-INVITE between two SIP phone to avoid having the media going through the server? Tks, Al __________________________________ Do you Yahoo!? Yahoo! Hotjobs: Enter the "Signing Bonus" Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus

thanks for the reply. the problem is after i start icecast with icecast -c /path to icecast.xml, i see the following: Changed groupid to 501. Changed userid to 501. and then it just hangs. i do not get a command prompt or anything. so i was wondering if there are any other messages that normally follow during the startup process. <p>--- Karl Heyes <karl@xiph.org> wrote: > On

Just spended ~ hour googling - all boards are based on GC-XX or I750X Chipsets - all for Xeons. There also some boards for Pentium 3. Can someone point me to a board with 64Bit 3.3v PCI for ordinary P4 with 800Mhz FSB. Thanks

Hello! Has anyone gotten 4.9R to recognize the "ServerWorks CSB6" ATA controller on a Dell 600SC? How about installing 5.2R on a 600SC? 5.2R recognizes the said controller, but it corrupts data. -- eof

Hi, I have a problem of loss of file ownership with rsync. For some odd reasons, I have to use a windows machine as rsync server, and to backup/restore a filesystem from/to a linux client. All files in my linux filesystem are owned by root, with gid root. If I save the files onto the windows rsync server from linux using rsync, and then restore them back to linux, the ownership for all files

Hello, What do you recommend for keeping track of user activities? For preserving bash histories I followed these recommendations: http://www.defcon1.org/secure-command.html They include using 'chflags sappnd .bash_history', enabling process accounting, and the like. My goal is to "watch the watchers," i.e. watch for abuse of power by SOC people with the ability to view

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

Hi, I'm running a simple test from asterisk towards a public telco switch (AXE10) over E100P. Here is the test case: 1) 30 calls are setup simultainously, 20 sec ringing time. 2) no calls answers (just calling a vacant public tax office :=) 3) Each channel will continue on its own with the same proceedure: ring for 20 sec, then hangup, ring for 20 sec, and so on. Of course this leads to

I keep getting the following Auto-congesting message whenever I try to dial from an X-Lite SIP phone to another one within my LAN. It's a real basic configuration but I am unable to figure out what is happening: localhost*CLI> -- Executing Dial("SIP/jay-de1b", "SIP/jtest|20|tr") in new stack -- Called jtest May 2 11:47:58 NOTICE[1133742896]: chan_sip.c:1019