similar to: Asterisk 1.2.0-beta1 Released

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

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

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, 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'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 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

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, Is there any future possibility that Asterisk will be compatible with connection to 3G video mobile phone such as Nokia 7600, Nokia 6630 and many ohters in Japan, Europe and HongKong? If this become possible, H.323 video clients and 3G mobile phone will be able to share video conversation, which will be huge in those countries. In Japan, more than 3 million 3G video mobile phones are

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

Due to a packaging error, the tarball released on Friday night did not have a version number embedded in it, which results in various strange build errors and other odd behavior. The tarball on the FTP servers has been updated to correct this situation. Sorry for the inconvenience :-)

Guys. I was checking the changes for 1.2.0 beta1 and I read this: * Asterisk Realtime Architecture * Asterisk Manager Interface * Asterisk Extension Language * Dialplan functions * More powerful dialplan expression parser * Portability enhancements for FreeBSD, OpenBSD, Solaris and Mac OS X * ... and many more! Can somebody explain a bit mor ethe part regarding

Hi, I've just installed Asterisk 1.2.0-beta1 for the first time and downloaded chan_capi and compiled it in and run it. (For comparison purposes, I've tried this with both chan_capi-0.4.0PRE1 and chan_capi-cm-0.5.4 and chan_capi-cm latest CVS). Whilst most things are fine, it seems that if I specify the 'b' parameter in the dial string before the number, sometimes the early

We're having issues with the FXO/FXS ports on our Digium TDM cards sporadically. I'm wondering if anyone else has had these problems, or if anyone can provide guidance diagnosing or fixing the issue? The symptoms are that the FXO and FXS ports "stop working", usually after 2-4 weeks of server uptime. When this happens, sending a (SIP) call to an analog phone on an FXS port

I have 1 x100p. Caller id works fine with the beta1 release. Cvshead releases fail with a combination of checksum and ss_thread errors? I'm concerned when beta2 or the 1.2 release comes out it will not work. I have been through the configs I can't find and changes that need to be made to get CVSHEAD to work. Thanks John Hill

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

http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta1.tar.gz http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta1.tar.gz.sig Now that v2.0 is in beta stage I don't expect anything big to change anymore. The configuration and the APIs shouldn't change in non-backwards compatible ways. There are still bugs left to be fixed and some small features to be implemented, but I'd like to

http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta1.tar.gz http://dovecot.org/releases/2.0/beta/dovecot-2.0.beta1.tar.gz.sig Now that v2.0 is in beta stage I don't expect anything big to change anymore. The configuration and the APIs shouldn't change in non-backwards compatible ways. There are still bugs left to be fixed and some small features to be implemented, but I'd like to

Hmm, this is peculiar! This mailbox has been unchanged on my test server since September. I've been using it to test each new version of Dovecot just after I've compiled it. When beta1 was released, I opened this mbox with no problems. However, when I upgraded to beta2, I got "file size unexpectedly shrinked in mbox file /export/mail/a/24/vis99003/INBOX (16895967 vs 16899267)".

I'm getting a FLOOD of these types of messages on my MAC OS X box: Sep 9 14:46:31 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:37 NOTICE[17627]: res_musiconhold.c:493 monmp3thread: Request to schedule in the past?!?! Sep 9 14:46:43