similar to: Announcement to called party

On Thu, Aug 14, 2014 at 12:34 PM, lvqcl <lvqcl.mail at gmail.com> wrote: > Jose Pablo Carballo <jose.carballo at ridgerun.com> wrote: > >> - channels = 2; >> - bps = 16; >> + channels = ((unsigned)buffer[23] << 8) | buffer[22]; >> + bps = ((unsigned)buffer[35] << 8) | buffer[34]; >> total_samples = (((((((unsigned)buffer[43] << 8)

Hi, In the last days I've been taking as reference the example found in examples/c/encode/file/main.c. With it I've been able to encode a 2ch, 16 bps, 44100 sample rate input WAV file to a FLAC file. Now I've been trying to modify this example to encode a 2ch, 24 bps, 96000 sample rate WAV file. I have to say I'm a bit lost on how I should read the input file in this case, and

I have 2 TDM04b cards currently running in an asterisk at home box that I am ready to replace with the CVS version of asterisk. What I am looking for is thoughts / recommendations. I want to move this to a small form factor ( shuttle ) machine and was wandering what expeience / advice there was for this? I have seen the incompatible motherboard list at digium ( and in fact I think my current

Hi All, Any solution on how I can implement prepaid billing on asterisk? But not the calling card type, just a simple Custome rwill buy credit, consume then buy again. Also, is there a solution for that when you combine asterisk with ser? Regards, Ronald

Hi list, Does anyone know the legalities of connecting an Asterisk box to the PSTN in Manila or where I can find this info out? I know it is illegal in some countries. thanks -Matt

This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream "Your call may be monitored or recorded, please hangup if you do not agree...etc" in a loop until the line is answered. Caller doesn't pay a single dime to

> -----Original Message----- > From: David Liu [mailto:david@deltapath.com] > Sent: 31 January 2005 14:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Announcement to caller when > called party haspicked up - without initial Answer()? > > > This is super easy to do. All you need to do is to put that > announcement

I checked the list of what not to use, but am still having no fun trying to find a working box. Can someone suggest a Compaq or Dell or MPC or . any other brand and model that is known to work well with the TE411P ? Will an old Proliant do? -------------- next part -------------- An HTML attachment was scrubbed... URL:

With the lack of info on Yoda Communications in Taiwan and their hardware, I thought I'd post my experience. I got my hands on a few H.323 VG-400's and VG-100TA's. http://www.yoda.com.tw/model.php?type=VoIP_Solution&pname=VG400 2 of the VG-400's were 2FXO/2FXS models. A couple were deployed to SE Asia, where we planned to offer our services. Originally, I ran a GnuGK server

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Rich Adamson > Sent: 08 February 2006 08:41 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update > ornot? > > However, if you expose the box to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hey all, First of all, hello again! Been a while since I've posted to the list, but I've been here lurking and watching ;-) Anyway, I wanted to pose a general question to the list to see if it turns up new suggestions for everyone/me. What is your preferred editor when coding in the dialplan? This is mainly aimed at those of you who write

Does anyone know of a way to make a standard analog phone plugged into an FXS port do something other than get a dialtone when you pick it up? For example, if the phone should automatically ring someone or play a greeting when picked up without having to enter an extension? - Robert -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have placed the custom-cdr-V1.0.tar for download http://www.itsngroup.com/software/asterisk/downloads/ Thanks > Dear Ben, > I've also the problems as Chris Mason, Can you send us your own AGI CDR > logging application? > Best regards, > Jian Hong Guan > France > www.directcentrex.com > > >

I would love to run Asterisk on an old laptop, in a mostly solid state configuration, with no HD. The laptop is slow (Pentium 233), and I need PCMCIA support (for my network card). Are any of you aware of a live CD that might work? Thanks, Dave

Hi All, I want to setup the interconnectionm between two servers, both having sip clients behind firewalls. I want the calls from any of the servers to land on any of SIP clients on the other. I am looking for dial out plans with the sample configuration files . Thanks,. satish

Is it possible with asterisk to setup aliases for SIP? For example, direct sales@mysipdomain.com to 55544@mysipdomain.com If this isn't possible directly with asterisk, does SER offer anything along those lines? A search of the usual sites didn't turn up anything conclusive. Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC http://www.djhsolutions.com

I've got a Mobile-to-PBX gateway installed and I want the ability to dial from my mobile phone into my PBX and next dial a land-line from the PBX so I can make cheep mobile-to-land-line calls while on the go. I've contemplated using the WaitExten application but it only seems to wait for ONE digit! Is there a way to put the calling mobile phone into a context and wait for a full-length

I know there was alk about this before but I cant sem to find it. Anyway to call some one and then play an IVR where they can make choices based on DTMF ? Thanks. Dovid __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

I have some troubles with ASTCC. TOOOOO often the "in-use" flag remains set. I would like to find a solution, where astcc.agi checks automatically if THIS user is in a call rather than to check the flag. If that is not possible, I would like to have an extension to dial to, and it will after hang up, reset the flag! The in-use flag remains set, if the caller hang up before the

Hi Guys, Im configuring my asterisk box as a voip gateway. I have TE110P which is connected on my PBX and i will be using voip for my outgoing. Here's my config zaptel.conf: span=1,1,0,ccs,hdb3 fxoks=1-32 zapata.conf: context=default signalling=fxs_ks group=1 channel =>1-32 -- Regards, Mark Quitoriano, CCNA Fan the flame... http://www.spreadfirefox.com/?q=user/register&r=19441