similar to: OT - Experience using Gmail for Asterisk Mailing List

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.

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

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

Hi Most of the threads in the list archive relating to X100P and hangups are about not detecting hangups. We have got the opposite problem. We have experienced an increased number of false hangups when connecting an X100P to an analog port of an ISDN terminal adapter. It happens more frequently on incoming calls than it does on outgoing calls. Often hangups occur after about 3-4 minutes into the

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

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.

Hi we've got Asterisk CVS-HEAD 18-Aug-04 (modified by Voicetronix as available on their site for use with the vpb driver) and an OpenLine4 (4xFXO). The same server also has two X100P. Calls on the Voicetronix card drop instantly when the called party picks up. The vpb driver reports that it detected a hangup (loop drop) yet there is no hangup when connecting the X100Ps or analog phones to

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

Hello, I'm researching the possibility of using VOIP (SIP) with an existing PABX system. Ideally, the setup would be to dial an outside line through the PABX (that would actually link to the the VOIP gateway). At this point I would prefer not to purchase a hardware-based VOIP gateway. I would prefer to use a software-based gateway for research & testing purposes. Could anyone please

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:

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)

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

A better explanation can be found here... http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0 > -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question

Hi there, I have an X-Lite phone on my box and I'm trying to register it with a remote Asterisk box. Both the X-Lite and Asterisk are behind a NAT. I know it's a pain to do because of SIP not working well with NATs, but I know there are ways to do such a thing...moving the Asterisk box outside the NAT is not a possibility at the moment. One thing we tried was setting up a VPN, but

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 need a cheap simple adaptor for analog phones to use with Asterisk. It should be some kind of "configure and forget" type of device, to use at the office, or just throw it in a road warrior's bag and use it while travelling, to call back to the "mothership". I can't decide between iaxy and sipura. Can you guys help? Which one would you use? (and why?) I feel that iaxy

Sorry to just jump in on the list like this, but I'm in a hurry. I've become lost in the mumbo-jumbo of the Wiki. Is there anyone on this list who would set up a Asterisk system for me? I have a Fedora Core 2 fresh install on an HP Pavilion 6553, along with a single POTS line. I'll buy the hardware I need to add to it to get Asterisk to work. We also have a couple of computers,

Hi, I was wondering if anyone knows where i can find out the standards used by telco's in different countries... and how to configure asterisk to support them. Secondly, whenever i try Dialing a zap channel, all i get is no sound on the phone source, and noise on the destination line. thanks Imran

Dear All I have successfully used the packaged version of * on the Mac for some time, but decided that I would recompile one of the more recent builds so that my PC and Mac were in sync. As suggested, I installed the XCode tools, updated bison and downloaded the latest version of *. Unfortunately, when compiling there are lots of errors, many of them relating to the non-existent /usr/src/linux