similar to: Best way to detect unknown and/or private incoming caller-id?

I have always been able to block toll-free numbers by catching them with a line similar to this for each DID I have on my system: exten => 5554441212/_888NXXXXXX,n,Playback(GoAway) Where 15554441212 is one of the DIDs that rings into our Asterisk box. The problem with this approach that I have to create a line like this for every pattern I want to block multiplied by every DID on my system,

Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument "IRQ" isn't numeric in numeric comparison (<=>) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci:0000:04:00.0 wcte12xp+

Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo cancelation and a quad FXO card. We have 4 analog lines, one of which is a Cellphone line for least cost

Hi @ all, i set a server to a costumer of mine with a TE207P for use with 2 E1 Lines. I set them together into one group in zaptel/zapata.conf The point is now, the customer has a free-volumina of 60k minutes per month, per line. How can i make a kind of load balancing, that both lines will be trafficed the same way ? I read something about DIAL(Zap/r1/.) for using round robin, and

I have spent some time this morning trying to add an Astribank to our current Asterisk, but it failed, so I just removed the hardware, restore the config files to the original setup and started asterisk.; I could see that no Zap channels are started so I did load chan_zap.so: pbx*CLI> module load chan_zap.so [Feb 26 10:32:18] WARNING[3809]: pbx.c:2968 ast_register_application: Already have an

Dear all, I have small question in sip.conf I added [service] type=friend ;username= ;secret= qualify=900 host=X.X.X.X dtmfmode = rfc2833 disallow=all ;allow=g729 allow=gsm allow=alaw allow=ulaw and I can proccess incoming call from soft phone only I calling on number that is used in extensions.conf(in example below it is 1) exten => 1,1,Answer; exten => 1,2,Playback(hello-world,skip);

I am using GotoIf all over the place in 1.4.8 but for some reason, the following in my dial plan: ############################################################# exten => _1NXXNXXXXXX,1,GotoIf([${EXTEN} = "15554441212"]?100) exten => _1NXXNXXXXXX,n,Dial(SIP/provider1/${EXTEN},60) exten => _1NXXNXXXXXX,n,Dial(SIP/provider2/${EXTEN},60) exten => _1NXXNXXXXXX,n,Hangup exten =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all I am working in a new set up with Grandstream GXP-2000 handsets. I like those phone, but they lack a feature I need: the phone cannot be locked by the user. What I actually want is a user to be able to avoid someone else making calls from his phone without giving him access to SIP configuration access to the phone. i.e. let say I want user

In my blacklist I have: database show ... /blacklist/Manitoba : advertising ... [incoming] ; First, lets take care of telemarketers exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten => 4,n,Set(goaway=${CALLERID(number):0:2}) exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1) exten => 4,n,GotoIf($["${goaway}" = "V3"

Howdy, I've had tinc working great for over a year now. They just made some network changes at work and now I'm only able to make a connection for a few seconds(30) or so before I get a "Metadata socket read error". I wan't to think that's some kind of firewall timeout thing but I'm not sure. I have no control over the FW or network at work so I'm not clear on

I would like to get in contact with users/consultants who are or have worked with the Cisco phones and Asterisk to trade information. Cisco has reluctantly made SIP available on their phones and most of the information on voip-info and other wiki's appears to be reverse engineered. There is a wealth of information out there which is terrific. I have a client with about 40 phones

What I would like to do is have two identical * servers which accept registrations of sip extensions 4000-4999. If I define a rrDNS or LinuxHA then I should have load-balanced registrations. However, say ext. 4001 is registered on *1 and 4002 is registered on *2, if 4001 tries to call 4002 then I would like to do something like: - lookup 4002 on *1, try to establish a call if it's

I have a subroutine to block spammer by CALLERID(number) exten => 4,1,GotoIf(${BLACKLIST()}?blacklisted,s,1) exten => 4,n,Set(goaway=${CALLERID(number):0:2}) exten => 4,n,GotoIf($["${goaway}" = "V4" ]?blacklisted,s,1) exten => 4,n,GotoIf($["${goaway}" = "V3" ]?blacklisted,s,1) but I just got another spammer (automated calls) who rotates his

Dear all, I am trying to find a solution satisfying the below equations in R. Set up the problem 9 X1+ X2 + X3 = 2 X1+ X2 + X3 = 1 which is subjected to 0 < X1 < X2 < X3 < 2. I have downloaded the packages \'linprog\' and \'lpSolve\' but can not see how to solve the question. Thank you for your help. With

I'm running Asterisk 1.6.2.5 with chan_skype on a x64 linux platform. When a user calls from skype (not skype-in) to asterisk, dtmf (basically menus for a conference system) works just fine. But when a user from the inside (soft or hardware sip phone) calls out via skype-out dtmf doesn't work. I have tried setting the codec to alaw, and dtmfmode to all possible options (auto, inband and

Anyone have any ideas why DTMF would not work on only one number? Looking through the logs, anytime a button is pressed, this is what shows up: 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Exception on 9, channel 1 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Got event Dial Complete(9) on channel 1 (index 0) 2006-04-13 11:39:18 DEBUG[22441] chan_zap.c: Echo cancellation already on We

This, IMHO, is one of the more annoying bugs with the newer GRUB (which i assume is the bootloader you are using). Specifically the newer grub can't, won't boot from a drive other than the one grub is on. suggest you revert to grub 0.97 or, if any boot loader is or can be put on the drive with the 6.5 distro you can make that drive the boot drive in the bios, which is how i do it. i.e.

I've searched around and found a few similar situations where the phone will call out when using a Asterisk server but not receive inbound calls. My issue is a little stranger. If I call out from the phone then the phone will receive the next inbound call. The phone will not receive another inbound call until a call out again from it first. Any ideas? I am using SIP and am using the latest

I'm new to dovecot and sendmail and my server doesn't forward mails to 'dovecot deliver' at all. I was reading http://wiki.dovecot.org/LDA/Sendmail and this http://www.dovecot.org/list/dovecot/2009-June/040353.html from the mailling list but I could figure it out myself. I someone could point me into the right direction. I'm confused with the config for LDA. Basically what I

We are in the process of preparing to move our Asterisk server to a Digital T1 interface card instead of a analog card (via an Adtran which is now connected to the T1). I did a preliminary test the other day and hooked the T1 line up to the T1 card, bypassing the Adtran. This worked rather well I must say. The two issues I ran into are: 1) Caller ID is not working even though I enabled