similar to: 3G Video Mobile Phone

I'm pleased to announce the availability of CHAN_NMS - channel driver for NMS Communications (now Dialogic) hardware. The attached README explains it all. The project website is http://chan-nms.hosting.lv/ CHAN_NMS is an Asterisk PBX channel driver that supports NMS Communications [1] (acquired by Dialogic [2]) Open Access line of products, that includes E1/T1 boards [3] for PCI(-X), PCIe,

Hi, I'm now setting up a VoIP conference room using Asterisk. All the clients are SIP phone (to be exact, Xlite), number of clients that should be registered are around 50 and concurrent users are maybe 15 clients at most. So, basically I think I can handle the situation only with Asterisk. I'm wondering however, most of my clients are behind NAT of home router and using SER together

Hi, I'm a beginner of voip and just wondering the possibilities of *. Is that possible for * to handle jabber based voice chat IMs, possibly inter-connecting them to different kind of clients -say, H.323 clients- in meetme conference function? If I use SER together with *, is that possible? -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter

Dear Asterisk Users, I'm a Japanese and now configuring Voicemail. Now I need to modify the way cmd VoicemailMain works to fix language difference and other my conveniences. What I want to do are... 1) Add words used in message retrieving guidance. I need to add different suffixes to numeric words due to Japanese way of mentioning time. (e.g. in English, you can say "Five

Hi, I'm going to setup Asterisk on my server which have 5 IPs (3 global and 2 local). Now I want to bind Asterisk to 2 IPs (1 blobal and 1 local) Is this possible on config? -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist http://www.macwebcaster.com

Hi, I'm thinking of setting up Asterisk for H.323 video phone clients. Now, what is the difference between native H.323 that come with Asterisk and "Open H.323 for Asterisk" ? TIA Kuni -- Kuniyoshi Murata.........................iChat/AIM:macwebcaster English-Japanese Interpreter mailto:kuni@ej-interpreter.net Macintosh Webcast Specialist

Hi, I'm using FedoraCore 1 for Asterisk 1.0. I assume that Asterisk accesses default audio device (say, /dev/dsp0) as audio capture device by application's default. (correct me, if I'm wrong on this) What I want to do is to let other audio capturing application (that is real producer, BTW) use Linux Box's default audio device. But, the default audio device is unavailable. Now, I

Hi, I'm thinking of introducing Asterisk on Linux for IP PBX. Now I'm using ISP that has VoIP service and I have VoIP terminal box for that ISP and a SIP account for SIP server of the ISP. Now, what I would like to do is the following. A. Setup IP PBX on Linux by using Asterisk. B. For IP-PSTN gateway, configure Asterisk to use my ISP's SIP account and connect to my ISP's IP

Hi, I know the following is mostly the issue of SER and I already posted the same content to SER User list. Just for more input, I posted it to this list. Sorry for the cross post for some people. I've set up SER for UA to UA call. I'm thinking of setting up SER to relay to Asterisk PBX to use conference call and voicemail of Asterisk. I will employ this system for client connection

Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing

Hello, I wrote earlier today about a new PRI installation in the Caribbean, where all outbound calls are functioning fine *except* calls to Sprint phone numbers, which get rejected immediately as "busy". The telco has been working with their switch manufacturer and took the output of "pri show span 1" from me and came back with this: ----quote--- Please check your timers

The first beta of Asterisk 1.2.0 has been released! It is available from the ftp.digium.com FTP servers, as well as the Digium CVS servers (under the 'v1-2-0-beta1' tag). This version of Asterisk represents a significant improvement in features, stability and compatibility over the 1.0.x releases. Some of the major new (or upgraded) features include: * Asterisk Realtime Architecture

iaxComm is an Open Source softphone for the Asterisk PBX. iaxComm compiles and runs on Win32, Linux and Mac OS X (Panther) systems. Recent Changes: * Improved jitterbuffer code * Steve Underwood's Packet Loss Concealment Code Features Include: * iLBC support * GSM support * speex support * ulaw and alaw support * Blind Transfer. * Custom Ringtones per

Hi guys I have implemented xapian on a website, and it currently has about 2M items in its index. Its all been working quite nicely so far, until I tried removing some old items from the index (removing items when the index was smaller was no problems at all). When I try to remove them now (using writabledatabase_delete_document() via php), it halfway freezes up the machine, and the apache

Hallo! I'm a part of a LAN, and I have samba running. My Linux server has no account in the windows PDC. How can i see the ohters wins clients and server? thanks

CRAN (and crantastic) updates this week New packages ------------ * disclapmix (0.1) Maintainer: Mikkel Meyer Andersen Author(s): Mikkel Meyer Andersen and Poul Svante Eriksen License: GPL-2 http://crantastic.org/packages/disclapmix disclapmix makes inference in a mixture of Discrete Laplace distributions using the EM algorithm. * EstSimPDMP (1.1) Maintainer: Unknown Author(s):

Hi We need to evaluate some open source project that supports 3G-324M on top of Asterisk. What do you recommend ? What has been your experience ? Thanks. regards, Anita Hall, Simmortel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100729/b7fcd340/attachment.htm

Hello, If anyone has successfully compiled asterisk with: app_rtsp codec_amr mp4_play mp4_save app_transcode h324m_call Please share the versions of OS software, and libraries used. Lets make this thread useful so that all tried and tested video resources of asterisk can be found in one place for ease of access and later reference. Thanks. -------------- next part -------------- An HTML

> > > > We may need some additional info. What kind of additional info? > I haven't put a ton of thought into > this, but I'm hoping we can either (a) use debug info as is or add some > extra (valid) debug info to support this, or (b) add an extra > debug-info-like section to instrumented binaries with the information we > need. > I'd try this data