similar to: Hard phones that support ILBC

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)

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.

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.

Aloha, Anyone know how to disable call forward on a Polycom Phone. Calls being accidentilly being forwarded somewhere is the #1 trouble that we have to respond to. The real issue is the 'end call' button becomes 'forward' when the call ends....therefore the user thinks they are pressing 'end call' but the call ends just before they press the button so they end up

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

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:

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

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

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

Hi for those who are unhappy with whatever mail reader arrangement they have reading the mailing list, I'd like to share my experience using Gmail, which I have been using for about a week or so now. I find Gmail to be excellent for the mailing list. It doesn't feel like a web mail application at all. The threading works perfectly. Responding to the list keeps the threads intact. It

Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML & can get input from the keypad & softkeys. Aloha, Matt

...now, we don't wish them ill, but Vonage seems to have been out of commission for quite a little while. Website is excruciatingly slow, log-in fails, hard-line and soft-line are out (alert-tone or "subscriber not in service") -- even "Network failover" is failing. Outgoing calls return fast-busy. 866-support number is busy. Let's see how they deal with that in

I am using IAX soft clients (firefly, IAXComm, IAXPhone) from a Win2K machine on a NATed private LAN to connect to my IAXTel account. All of my calls to outside numbers(1-800, 1-877, etc) seem to be disconnected after 9 seconds. Has this limit been placed on IAXTel? My call to the echo test number at 1-700-999-9613 seemed to run for as long as I wanted. --

Hi; I've just had a couple of discoveries in the learning process that are really making me impressed with this software. 1) IAX transfer - I'm running Asterisk boxes A, B, and C. B is in the middle and has a dialplan that points to extensions on C. When a client on A in the proper context on B tries dialling a client of C, B is smart enough to release itself somehow from playing