similar to: Not Hearing hello-world Play

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 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]: ***

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

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

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 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 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

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

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: > > >

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().

Basically, pretty much all the 2 line cordless systems I've seen come with a built in digital answering system that I'll never use, the main problem with this is that these units don't support VMWI (visual message waiting indicator) with telco supplied voicemail. This is a problem because I'll be setup an Asterisk system in the next month or so to handle my 2 analog and 1

Can anyone recommend a good cordless user-configurable SIP hardphone that is readily available in the states and doesn't cost $300? There seem to be a plethora of decent and affordable corded phones (like from Grandstream) but the search for a cordless unit seems elusive. I purchased a vtech 8100 online only to discover after receiving it that it is locked to vonage service. Thank you.

I am currently looking at putting together an Asterisk PBX system for my home office. One of the things I need is an expandable 2-line corded/cordless phone system. If I wasn't using Asterisk, I'd be getting one of these AT&T systems with 2 extra handsets. http://telephones.att.com/attui/shop/product_detail.cfm?itemID=ad53fe08-495e-41b6-b390-3aaa1c84cc40 So, the question is,

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 list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

So, when I get no comments on this at all, either here or on any of the forums, does that mean nobody knows what I'm talking about?? Or does nobody know the answer?? Or is it just a stupid question and nobody wants to bother telling me where to look?? It *is* a question that I have to answer somehow; I've read all through TFOT and see nothing relevant to this issue. It's silly to

I have a dilemma. I am trying to setup an asterisk setup for about 10 people, but with the ability to expand to 100s. We are looking into hardphones to use for our systems. Here are the requirments I am looking for: SIP Not POE Full duplex Speakerphone $100-$300 We would like to have the cordless and corded option if possible (at least for 3 of the 10 people). So here are the