similar to: Remote UNIX connection disconnected over and over

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).

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

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 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

I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better!!!! I could use a different context for this customer and use only his account code as card number. Any advice would be

Hi, I'm interested in anybody that is providing a phone support service using an Asterisk system, with built in charging system. I run a PC support company and use Asterisk at the home/office. I would like to be able to provide technical support to my customers using asterisk. However I want to be able to charge them fairly for this support, and with little work on my part. My idea was for

I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data package. I continue to pay for a standard voice line in addition to the broadband connection only for local calling, fax and emergency 911 use. The way it works is that the fiber optic connection is terminated on the house via an ONT (optical network termination). The ONT can provide connectivity for three types of

-----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

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