similar to: Packet flow in relaying from SER to Asterisk

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

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 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 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, 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

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

Dear R users: I have an elementary question: how to creat a vector of [A1, A2, A3...... A300]? I know c(1:300) would give 1, 2, 3, ...., 300 but not sure how to attch a A to each element. Thank you Yihsu Chen The Johns Hopkins University

I have installed Asterisk on a CentOS4 box and then installed Asterisk from CVS. I installed a Sangoma A101 and connected it to a Adtran 600 using a T1 Crossover cable. The 600 has 12 x FXS, 12 x FXO interfaces. I ran through the wanpipe install instructions and configured it, now I can run [root@altpbx asterisk]# wanrouter hwprobe ------------------------------- | Wanpipe Hardware Probe Info

Hi, When I run puppet agent --test --environment 42A, I have the following warning : Warning: Local environment: "42A" doesn''t match server specified node environment "production", switching agent to "production". ... The puppet manifest for the environment "42A" isn''t applied. The puppet version is 3.3.1-1puppetlabs1 on agent and

What is the best free stun server out there? The one that I have looked at from vovida requires two NICs. Is this neccessary?

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type"> </head> <body bgcolor="#ffffff" text="#000000"> I have an Asterisk 1.4.18 with a mix of cordless phones connected using Linksys SPA2102 ATAs and Cisco 7940G

After running several days in the same configuration, my 3.0.6 nmbd crashed with the above error (see log below). Maybe there is a connection to this message some seconds before: register_name_response: server at IP 192.168.0.4 rejected our name registration of AK<1d> IP 192.168.0.12 with error code 6. The ip .4 machine is a win2k server, which works (or should work) as an ordinary

Hi @all, this is an very old problem, posted the first time 2005 to samba-testers at samba.org: If I have a Domain client with WinXP, which is part of a domain and using this client, if the PDC is not available and want to remote shut down this client with the net tool from other servers, then this is not possible, because net tries to authenticate against the PDC: testeis # net -d3 -S

hi list, We got the following problem: We try to use samba 2.2.5 as a printserver for windoze clients (samba 2.2.5 on RH linux 8 w/ CUPS 1.1.15). CUPS works w/o any problems, but when we try to generate the drivers for the windoze clients (by using cupsaddsmb), we get the following error: >> cli_pipe: return critical error. Error was SUCCESS - 0 result was NT_STATUS_UNSUCCESSFUL

Hi All, When I try to do "cat message.txt | smbclient -M <machinename>" or "smbclient -M <machinename>", I get: added interface ip=192.168.0.100 bcast=192.168.0.255 nmask=255.255.255.0 <snip all other interfaces; about 25 more> Got a positive name query response from 192.168.0.12 ( 192.168.0.12 ) session request failed What does "session request

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)