similar to: Overlapping Playback() with Dial()?

Hi All, I have a telephony project for which I need to build a prototype to demo for management. The prototype must work on a GSM phone network. In the demo system, a call from GSM phone comes into the demo box. The demo box runs CallWeaver. Callweaver picks up the GSM call, answers it and plays a sould file, then dials out to a second GSM phone somewhere and connects them so they talk. My

Hi All, I'm compiling zaptel. Did the usual ./configure, then make. Compile breaks saying: ---------------------------- /usr/src/zaptel-1.4.5.1/wcusb.c:1451: error: unknown field ?owner? specified in initializer /usr/src/zaptel-1.4.5.1/wcusb.c:1451: warning: initialization from incompatible pointer type make[3]: *** [/usr/src/zaptel-1.4.5.1/wcusb.o] Error 1 make[2]: ***

Hello My Aster-Friends! I would like to hear if anyone out there in Asteriskland has used the Dock-N-Talk (DNT) box to connect cell phones to Asterisk box. I have a couple of these boxes that I need to make work with Asterisk, connected with Digium TDM400P card. Anyone tried it before, and how did it go? Thank you. Jeng ___________________________________________________________ Yahoo!

Hi All, I was wondering, what tools are readily available out there in Asteriskland for me to use in stress/load testing asterisk box I have in the lab. I want to observe how my box holds out under heavy/light/medium load. Thanks, Jeng ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good

My Gurus! I'm still playing with asterisk in the lab here. There is a feature that I need in a production asterisk system. I was wondering if it already exists in asterisk. When we want to shutdown a production asterisk system, we would like the shutdown to happen after there are no more calls being processed. In other words, a shutdown command that does the following: - block asterisk

Hi Asterisk Gurus! My lab asterisk box has 1 FXO and 1 FXS ports in it. I connect a GSM phone to the FXO port. I connect a regular corded phone to the FXS port. The Dial() application for both incoming and outgoing calls specifies the A(hello-world) flag. From another GSM phone, if I call the extension (corded) phone attached to the box, it plays the hello-world file when I pick it up. But

described (stop accepting calls and shut down when all calls have completed). If you don't want to stop accepting calls, but still want to stop Asterisk when there are no active calls, you can use "stop when convenient". The same qualifiers ("gracefully" and "when convenient") can be applied to the "restart" command. Cheers, AR On Dec 10, 2007 7:29 AM,

Hi People! Is there an underground asterisk command reference manual that the Gurus here share amongst themselves only? :-) The reason I ask is that sometimes I see mention of an asterisk command and I scramble for my asterisk book (pdf) to look it up but can't find it in there. For example, I saw here last week people talking about the Set() application with the "If" conditional

Hi Gurus! Please excuse this pesky Asterisk rookie....:-) I just wanted to know which channel variable tells asterisk the number of rings before an incoming call on FXO channel is answered? I looked through zapata.conf.sample and other places and could not find something there readily. Thanks, Jeng ___________________________________________________________ Want ideas for reducing

Hi All, The Asterisk book (the pdf version) is excellent!! I want to thank all the guys that put it together. I am most grateful for it. There is something about writing a dialplan that I'm not clear about. What I'm trying to figure out how to do is this: when I transfer a call to the destination number, and the person called picks up the call, I want to play a greeting message to the

On Dec 10, 2007 10:36 AM, Jeng Yu <jengyu2007 at yahoo.co.uk> wrote: > Thanks, All! And thanks, Oquendo! I will experiment > with this suggestion. I was actually thinking in terms > of a situation where it would be done > non-interactively. > > Jeng > > > --- "J. Oquendo" <sil at infiltrated.net> wrote: > > > Jeng Yu wrote: > > >

My Esteemed Gurus! Still learning... I need to read about and learn how to configure Asterisk box here in the lab for digital cards. I'm about to get one of those E1/T1 interface cards from Digium. Please point me to docs that I can read so I don't have to bug you with too many questions :-). Sample config files would greatly help. In particular, suppose I have a T1 card in my box

Hi Friends! I need help! I'm still Asterisk rookie, so please forgive me. My Asterisk is no longer answering incoming call on the phone line. I call the phone and it rings but asterisk is not picking it up. The phone line is attached to port 4 (FXO) on my digium TDM411P card. I am running Asterisk 1.4.11 with zaptel-1.4.5.1 and libpri-1.4.1 on Fedora Core 5, Linux Kernel

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hello, I'm using wine-0.9.35, and when running an application Wine tells me to install the Windows version of Mono. I have done this (version 1.2.3), but it still fails to run. The relevant error is: Unhandled Exception: System.TypeInitializationException: An exception was thrown by the type initializer for System.Windows.Forms.ThemeEngine ---> System.ArgumentException: FontFamily

Hi, I am looking for AEC software which can be run on the server side. This means there will be a fairly constant 600ms or so gap between sending out an audio frame and getting it back with echo. Could Speex AEC be configured to handle these conditions? If so, how good can I expect it to be? Thanks --------------------------------- Yahoo! Mail is the world's

Hi, I am trying to set up a CentOS 4 machine with various iSCSI mounts for Oracle, but devices keep on moving around, causing the mounts to fail while called by the sd[a-f] names. We thought about using labels, but, as some of the partitions do not use EXT3, that was discarded, so we thought about using udev and by-id. The problem is that udev does not seem to create the by-id folder or any

Hi, I am trying to set up a firewall rule so calls to old_mailserver:25 get redirected to localhost:25. I have seen quite a few rules and none seem to work. I have tried with the firewall enabled (configured to allow smtp) and disabled, but it doesn't seem to make a difference. One of the commands I have been using is: /sbin/iptables -t nat -I PREROUTING -p tcp -d old_mailserver_ip/32

Hi Jean-Marc, Regarding you points: 1) Is it ok if the audio is encoded (using Nelly Moser ASAO) and sent to the client and decoded when it is recevied so the AEC is always performed on raw PCM16 8KHZ ? 2) The audio is moved in 32ms (512 byte) chunks and the reading and writing to the AEC code will be done by separate threads at regular 32 ms intervals. 3) Occasionaly audio is

The client is the adobe flash player. No install and on 98% of all desktops but we can't change it. It works ok if people use headphones but we need to stop the howl than can build up if more than one person in a conference has mic to close to speakers. Any ideas? Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> wrote: > 1) Is it ok if the audio is encoded (using