similar to: SER is a better NAT solution?

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 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

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

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

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 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

I have some data I would like to model which involves choice of food by dung beetles. There are a number of experiments where in each case, there are five choices. Overall there are more than 5 different foods being compared (including a placebo) and different experiments use different comparisons. The problem is a generalization of Bradley-Terry but it differs from some generalizations in

Hi! Please help me... I'm really confused.. I have read almost all the books out there but its not working.. I just want my Windows PC to be able to access UNIX PC with a username and password authentication. Below is my smb.conf file.. [global] workgroup = MyWorkgroup netbios name = board_pc server string = %h server (samba %v) log level = 10 syslog = 0 log file =

Hello, On July 22th, 2010, Typo version 5.5 named for famous photographer Richard Avedon was released to the public. Coming with a new admin and setup, Typo 5.5 runs on Rails 2.3.8. Being around since March 2005, Typo is (probably) the oldest blogging platform in Rails. It has a full set of feature you would expect from such an engine, powerful SEO capabilities and full themes and plugin

Hi! Please help me... I'm really confused.. I have read almost all the books out there but its not working.. I just want my Windows PC to be able to access UNIX PC with a username and password authentication. Below is my smb.conf file.. [global] workgroup = MyWorkgroup netbios name = board_pc server string = %h server (samba %v) log level = 10 syslog = 0 log file =

Hi Dudes, i searched for some time for an answer for this, i found some posting from John Todd half a decade ago [1], was there some chance in this? Is it somehow possible to voip from ichat to asterisk? If there's no light, is this something that could happen with enough founding, or is "Mapple" preventing this somehow (legal or technical)...? Regards, Andreas [1]

I've been monitoring this problem for almost a month now. I realized that it is more extensive than what I described previously. I can very easily replicate this problem on every Zap channel. Following is the senario: 1. Call Zap/5 via say SIP/15 -> Zap/5-1 created and starts to ring 2. Call Zap/5 via say SIP/21 -> Zap/5-2 created and starts to ring 3. Hangup SIP/15 ->

I have an ext3 filesystem that I want to mount without loading the journal. I tried the "noload" option with both 2.4.9 and 2.4.18 series kernels and get the errors listed below. [root@host]# mount -t ext3 -o noload /dev/sdf1 /mnt mount: wrong fs type, bad option, bad superblock on /dev/sdf1, or too many mounted file systems /var/log/messages: Apr 18 13:30:23 host kernel: ext3:

Hello Alex, We have a customer looking to replicate OnSIP using OpenSER/Asterisk or FreeSwitch. Can you provide us a quote on the cost to completely replicate OnSIP? Thanks in advance, Ed Direct: 678.522.8511 Mail: edpimentl[at]gmail.com] Voip/IM: edpimentl [SKype | GoogleTalk ] -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear I am using ser + asterisk, for setting up land line calling. only probelm, each unregistered soft phone can places the call only with callerid, this is critical problem, because any number(soft phone) , has a limit time to use this system, best Mani ____________________________________________________________________________________Be a better Globetrotter. Get better travel

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <font size="-1">Hi,<br> <br> i use a ser, as proxy sip server(authentication), then a cisco router as sip2h323 gw(authorization and accounting). i want to start asterisk as sip statefull