similar to: RE: Random disconnects

I have just installed * (RH9, P4 3.0GHZ, 1G RAM) in a small office, using three TDM400's with 4 FXO's each for incoming calls. Outgoing calls are (for the moment) routed via VoicePulse. Phone sets are Cisco 7940G's using SIP. I'm getting intermittent echo on outgoing calls, and my understanding, based on reviewing the wiki and several posts here, is this: >>>> The

Hi, I am sorry if this message has been reposted, but for some reason I am having problems with posting it. I configured asterisk and zaptel modules with fedora2. I want to be able to load the zaptel wcfxo and wcfxs modules. For now I will use only the Wildcard TDM400P card. I am able to load the modules but I cant configure them using ztcfg or zttool because the tools are compiled to use the

I'm getting these "calls" at 16 and 46 minutes after every hour. The SIP phone rings, and if we pick up, we get a dial tone. If we don't pick up, we get the dial tone in a voicemail message. An analog phone connected to the incoming POTS line doesn't ring (whether or not * remains connected to the line). It's like the horror movie where the babysitter is getting

Hi all, I've installed Asterisk on Linux Red Had 9. Now, I was trying to set up a GUI based system for the PBX. I downloaded some packages, but I have to have Perl running CGI scripts through the webserver. It does not allow me to. I am able to run a basic script that just just prints out html messages and nothing else. However, when I try to run am-main.pl or config.pl or any other

So Asterisk can't send VOIP calls to Vonage -- but it is still possible to use Vonage for flat-rate long distance by connecting the Vonage AT-196 to an * FXS port, right? The price is an extra D/A <--> A/D conversion. Jim Shilliday IT Director Equal Justice Center 1315 Walnut St. Suite 400 Philadelphia PA 19107 215-238-6970 -----Original Message----- From: Tim Petlock

My users are reporting that some voicemail messages are being cut off in the middle of being played back. The recordings are OK (they play fine when forwarded to e-mail, and they can often be accessed OK during a later call to voicemail). I found nothing in the archives on this -- ideas anyone? RH9, P4, CVS-HEAD-09/02/04-08:44:34, aggressive echo suppression turned on. Jim Shilliday IT

Anyone have some good examples if passing a call from one switch to another using IAX. I would like to have a call come in over PRI and pass it into a certain context of another server. Bother server seem to register with each other fine... I thought the command was... switch => IAX2/user:secret@host/context but I can't get it to work. Can anyone shed some light? Thanks, Chris

Hello, I cannot accept any inboud calls from any provders in my asterisk which tries to authenticate the provider and at the end rejects the call with tthese message 407 Proxy Authentication Required How do I turn off this message. Thanks. Ehsanul Karim

This for the archives in case it may help someone: I was able to upgrade two Cisco 7940's from firmware P0030301MFG2 to SIP 7.1 as follows: 1. Installed 7.1 images from the Cisco zip file to the TFTP server. 2. Specified "image_version: P0S3-07-1-00" in SIP<MAC>.cnf and SIPDefault.cnf 3. For the older of the two phones, renamed P003-07-1-00.bin to P0S3-07-.bin, making it

Ming-Wei Shih wrote (Re: [Asterisk-Users] Best Linux for Asterisk) > I am running * CVS head on Gentoo/i586 > and Gentoo/Sparc64 (US60 2x450/1GB RAM), > they are running great. > > On sparc64 * does not compile out-of-the-box, > some hackings in the Makefiles are needed. Great stuff. Please, can you share your adjustments to the Makefiles with the community?! If you don't

Hi I had asked for some help with the Asterisk Assistants http://www.voip-info.org/tiki-index.php?page=Asterisk+Assistants+for+MacOSX and many have offered assistance with translations which I am grateful for and like to say thank you again. However, there hasn't been a single response from a graphic designer to offer help with a custom icon. Are there any graphic designers on this list at

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)

Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a

I am still haveing problems (echo) with my X100P but I'm thinking it has more to do with the server it is in which is not a negotiable item at this time. My question then is to the use of SPA3000's as a replacement from the FXO standpoint. 1. Can you setup the FXO port to recognize distinctinve ring and call a different context like you can do with Zap channels? Being able to call a

I have Asterisk server providing phone service for my company. The server is behind a PIX-515 FW and is assigned a private address 192.168.11.X/24. With that said what is best to provide remote SIP phones (home offices) securely. If the solution is to put up another Asterisk server with a public IP address I am opposed to that. I am looking for the a secure reliable solution to set up remote SIP

I'm working with the final details of the Astricon agenda. I haven't got anything so far on Asterisk GUI's and there are plenty of projects out there. I would like to invite developer's of Asterisk GUI's, both open source and commercial, to participate. What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI

Hello, Please be conscious of Digium's bandwidth and use a Mirror when downloading 1.0. I have mirrored the tarballs at: ftp://ftp.nacs.net/asterisk/ Direct links: ftp://ftp.nacs.net/asterisk/asterisk-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/asterisk-sounds-1.0.0.tar.gz ftp://ftp.nacs.net/asterisk/libpri-1.0.0.tar.gz -- Vice President of N2Net, a New Age Consulting Service, Inc.

Logan O'Sullivan Bruns wrote: > I know Solaris isn't a well tested platform and I did have to make > some minor code changes to get to compile on my sun box. Well done! We need more momentum for Asterisk on non-Linux platforms. Building a community around Solaris much like there is a community around BSD, would be very helpful. This will only happen if Solaris users start sharing

Is there a way to override the SIP From Header that is used in the extension.conf Dial command? The default is 'asterisk@host'. I do not want to configure SIP accounts in sip.conf, but instead generate the SIP From-User within extensions.conf from data the user has entered interactively. Any idea? Henrik -------------- next part -------------- An HTML attachment was scrubbed... URL: