similar to: Call and then play IVR

-----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

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

I am trying to send an announcement to the called party using the A(x) parameter in Dial, however, the message is not being played. There is a pause between the Dial command being executed and the call being connected to the calling party of the same length as the announcement .gsm file, but the message itself is not being played. (I have tried this and timed it with different announcement files).

<RANT> Funny your concerned about copyrights and moral issues regarding the work of others. One question you may want to ask YOURSELF is: Why would I use as my email a copyrighted work followed by the name of the Company that owns the copyright??? asteriskdigum@yahoo.com, Come on!! Who are you trying to fool? Are you out for the fast buck, by having someone that thinks you work for

In PHP something like: $pid = pcntl_fork(); if ($pid != 0) { // we are the parent // do parent stuff exit; } // we are the child, detatch from terminal $sid = posix_setsid(); if ($sid < 0) { die; } // do child stuff On 04/19/2019 02:00 PM, Mark Wiater wrote: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking

Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

Glenn, I can't use MySQL as each node currently has MySQL however there is a lot of data that is stored locally on each box. I may have to take this route if I can't find something else but that would mean syncing all sorts of data that does not need to be synced. On Tue, Jul 30, 2019 at 10:03 PM Glenn Geller (VDOPh) <ggeller at vdo-ph.com> wrote: > Hi Dovid, > >

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 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

Is there a way to terminate a ringing call before it is answered? I am speaking of prepaid card application in which you want to make another call, because the current number it is not being answered, and you don't want to hangup before dialling another number. /Obelix

I (cannot sleep and I) am thinking if there is a way to make inwards billing easy possible. To dial out we use something like: exten => _9011N.,4,DeadAGI(astcc.agi,${CALLERID(num)},${EXTEN:${TRUNKMSD}},${TARIFF}) (I have an extra field TARIFF, what allows me to use different prices for different users) To dial to a phone we use something like: exten => 888888888,1,Dial(SIP/6001,20,tr)

Hi, I am pretty sure that you already answer to this question, but I was not able to find the solution on the console I have over and over the following msgs -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection disconnected -- Remote UNIX connection

Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the