similar to: list proposal

Arrrrgh I hate trying to sift through all these messages and keep track of the various threads going on ......... Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers....... only problem being that this list requires list membership before

Supposed scenario: one PC(2GHz CPU), one card 4E1, and one Internet link. There is somebody he know (has experienced) how many concurrent call (Classical Phone->Voip) can handle Asterisk ? Thanks ! -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Hi Folks, have a question, on bandwidth. I want to run an asterisk server SIP to H323, g729. Calls arrive on sip/iax go to IVR get authenticated and egress through h323. So G729 license is only used during IVR and then it is pass through. I am collocating this server. Colo offer a monthly bandwidth quota. Lets say I want to do 100K minutes per month of VoIP calling at the beginning. What would

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 --

Just starting my search for a SIP/MGCP/H323 channel bank. I need analog ports in a building that only have network connections back to my * server. I could install another * server and use a normal CB with a PRI but I would like to investigate any good CB's with network trunk abilities Thanks Dave Packham

Hi all, I can see huge traffic here over 400 post in 4 days. My proposal is to create asterisk newsgrup proposal or phpBB forum what do think about it ? BR, Corvin btw. I'm admin of phpBB Forum (slackware forum - polish language), nearly 900 users. I think if someone will prepare it good it can be great project. (but I have 7 person team).

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

Anyone done a chat line app? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956

Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before?

Hi, I'm a student and my thesis work consist in testing Asterisk with Swyx(SwyxWare). My approach is to declare asterisk as h323 gateway for the Swyxserver using oh323 Plugin. Is there any possibility to connect Asterisk with Swyx? how? the outgoing call must pass from Swyxit->to Swyxserver-> Asterisk->to PTSN Thanks

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

Hi, i am recording a message, but it is always cut off at 40 secs. There are no time out configured. Gabriel -- The educated person is not the person who can answer the questions but the person who can question the answer.

I have set up a couple of test Asterisk servers and have never had a problem with sound. I've just done a fresh install on a dual 1GHZ PIII Asus box running Fedora Core3 with the Digium PCI Dev kit and following all the various Core 3 How-To's. I can make calls ok but when any sound is sent from the Asterisk box such as voice prompts and music on hold the sound is completely chopped up in

<P>hi everyone,</P> <P>I know this issue has been raised many times before, i think still the problem remains. When a call is made through a Zap channel, whether it is actually made or not (irrespective of whether, engaged, busy, or actually answered), asterisk logs it in CDRs as a call made. This makes it impossible to do an accurate billing. Has anybody found a way to overcome

Does anybody developed Predictive Dialer using Asterisk/Digium PBX? Another question: does anybody developed an Dialer using the X100P board? Julio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030718/be051d49/attachment.htm

Forwarded back to the list so others might get the benefit of the answers, and I get fact checked by others. -------- Forwarded Message -------- > From: Lee <leeb00@gmail.com> > Reply-To: Lee <leeb00@gmail.com> > To: Steven Critchfield <critch@basesys.com> > Subject: Re: [Asterisk-Users] udev or not? > Date: Fri, 10 Dec 2004 13:00:29 -0800 > On Fri, 10 Dec 2004

Can T400P be configured for nfas (one d-channel providing signaling for more than one span)? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030522/f2b637a9/attachment.htm

hi guys, do you have any suggestions on how to connect 2 TE410P via E1? (for simulation and testing purposes) asterisk1 --> TE410P ----> ? ---------> ? ---->TE410P -->asterisk2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030907/698cd499/attachment.htm

am compiling h323 support using channels/h323/ error am getting is this any pointers chan_h323.h:30: warning: `struct sockaddr_in bindaddr' defined but not used g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L/root/pwlib/lib -lpt_linux_x86_r -L/root/openh323/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat /usr/i486-suse-linux/bin/ld: cannot find