similar to: Before Bridging Two Calls

Hi, I have a production asterisk-1.2.8 system with FreePBX & PRI Digium card. I am looking for a paging system to an external speaker. I can page to internal Polycom 501 VoIP. But, what hardware or system do I need to integrate with the asterisk to have this acheived. -- Deepak Linux your Life, Don't Window it [[]] { All for the best }

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

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

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

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: > > >

Hi All, Can I overlap Playback() with Dial() in a dialplan? For example, I have this scenario: A call comes in, Asterisk picks it up, does Background(enter_number), then does Playback(bulletin_message), and while the Playback() is still going, I want to execute Dial() to the target extension so it overlaps with the Playback() and the call will be bridged instantly upon completion of Playback().

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

I am moving my first steps in writing and compiling C code and calling it from R. (I am also new to shared libraries...) My problem is that my C code uses a C function contained in another library (".a", not ".so" - is that the problem?). This is how I compile the file with the functions I want to call from R: cd /export/home/gpetris/ R SHLIB tryit.c cc

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

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine

The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain web browser specially VP8 codecs so a friend of mine

I'm testing dovecot as a replacement for an aging UW-IMAP server. I have the basic IMAP setup working but I'm stuck on public folders (which my users are addicted to). I set up public folders generally a described in the wiki. The problem is that when a user creates a new public folder, the folder is owned by the user with permissions drwx------ which prevents other users from

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

Hi ! My name is Gerald and I am working with WEBRTC and JsSIP.<div><br></div><div>I configure my Asterisk 11.7.0 to work wit WEBRTC.</div><div><br></div><div>Using a JsSIP (http://tryit.jssip.net/), the SIP extension can connect at the Asterisk, but when we try to make a call they send a 488 response and finish