similar to: VOIP providers

Hello Everyone, I'm using Mandrake 10 and trying to learng Samba 3.0.2a. I compile the source and install it alright on my Mandrake linux (./configure, make, and make install). Here what I'd done after the installation: Add to /etc/services file swat 901/tcp Add to /etc/inetd.conf file swat stream tcp nowait.400 root /usr/local/samba/sbin swat Since swat binary in the

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

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.

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

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

I set up an asterisk box with a broadvoice sip connection for incoming connections it works great when I use a cell phone, vonage line, calling card to call the asterisk box, but when I try to call it from our verizon land line it is busy and asterisk logs do not show incoming call. Any ideas on what the issue is? Thanks! Randy

I can cut and paste the log file from a reload right now, and provide you with the other information when I get home after work: tmp*CLI> sip debug SIP Debugging Enabled tmp*CLI> reload Mar 21 14:52:42 NOTICE[23231]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us' 11 headers, 0 lines Reliably Transmitting: REGISTER

Hi, M new with samba i just installed sambav3 and configure the sam.conf file to join a windows domain(win 2000). when i use net rpc bind to join this solaris bix it says joind the domain. but m not able to ping the solaris box from windows machine as well as the reverse. could you please suggest me if i need to do some other configuration.. Thanks in advance Regards Sids

Stealth Communications Announces Registry to Avoid Access Fees Posted on: 04/23/2004 Stealth Communications Inc. today announced the official launch of a registry that allows service providers routing calls over the Internet to avoid paying local phone companies access charges. The VPF ENUM Registry allows carriers to map telephone numbers to IP addresses for such things as SIP phones and

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

I've been trying to use net2phone's sip service at sip.net2phone.com with * but keep getting SIP/2.0 401 Unauthorized. Do you know if this should be possible? So far: I can use an ata186 to connected directly to n2p through sip.net2phone.com without any special settings. I can connect from * to iconnecthere, but, whatever I try from * to n2p produces "SIP/2.0 401 Unauthorized"

Hello, My server is Mandrake 10.1 eth0 is WAN with static IP connected to 512k DSL eth1 is LAN. I am using squid proxy for internet with NSCA auth. I am able to send and recieve mails. One of the client system wants to be able to make net2phone calls. As of now he is not able to. Howto allow net2phone calls ? Thanks Varun