similar to: 1.0 Mirrors

Greetings all, My next step is to purchase a nice VoIP phone for my desk. I have a grandstream, and the sound is great, but I'm looking for more of an office style phone, preferably that can handle multiple lines, has a more flexible display (i.e. name as well as number). SIP would be preferable. Any suggestions? Thanks, Eric

Hello, Straight from the floor of Astricon 2004, I am happy to release my updated Asterisk 1.0 RPMS for RedHat 7.3 and RedHat 9.0 platform. Current Release --------------- asterisk-1.0-0 libpri-1.0-0 zaptel-1.0-0 kernel-module-zaptel-1.0-0 RedHat 7.3 ---------- ftp://ftp.nacs.net/asterisk/rh73/RPMS/ ftp://ftp.nacs.net/asterisk/rh73/SRPMS/ RedHat 9.0 ----------

Hello all, It's my birthday today, so as my present I would like everyone possible to download and test my updated set of RPMS for Asterisk 0.7.1. By popular request, I installed and built a set of RPMS for RedHat 9.0, and in the process fixed a bunch of issues from the initial build. I have also updated and will be maintaining a page on the Asterisk Wiki located at:

Hello all, Per my last message to the list, and my promise to the Developers that I'd create RPMS if they released 0.7.0, I would like to announce the availability of experimental RPMS for Asterisk release 0.7.1. These are targeted at RedHat 7.3 systems, running the latest Kernel release (2.4.20-28.7). As the RPMS mature and people submit comments, changes, updates and patches, I will

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.

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

Hello, I'm looking at building a couple new PRI Gateway boxes using TE405P cards, and was wondering if anyone has had any experiences (good or bad) with the Intel SE7210TP1_E motherboards from Intel. General Technics builds some really nice (and cost effective) 1U servers based on the board: Server: http://www.gtweb.net/gt637.html Specs:

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 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

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

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:

Howdy, I just did an "apt-get dist-upgrade" on my Debian unstable box, and noticed that the Asterisk version appears to be 1.0-1 in the unstable tree. I KNOW that 1.0 hasn't been released yet, so I am wondering who is responsible for the Debian packages? This will be VERY VERY confusing for people and it should be corrected ASAP. -rw-rw-r-- 1 1176 1176 1189696 Jun 7

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)

Is there a user guide for Comedian Mail? I need to give some training materials to my end users. So far, I have been unable to find anything through google or the Digium site. THanks, Wiley The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other

hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am

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

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

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

Hello, At one of our locations, we have started to see Polycom 501s (running 1.6.7 firmware) randomly reboot. We have taken packet traces of the phones to determine if there is something odd in the Layer 2 or 3 of the network that might cause it, and have not seen anything strange. There are no errors on the ports. This appears to be affecting POE powered as well as AC powered phones. The Polycom