similar to: Blacklisting Toll-Free etc.

I am aware of how to match a particular caller-id or a caller-id pattern and do something with the call like this: exten => 15554441212/_888NXXXXXX,n,Playback(GoAway) What I am curious about, is the best way to block unknown, private and 000-000-0000 calls. I know I can do this for 000-000-0000 calls: exten => 15554441212/0000000000,n,Playback(GoAway) Is there a better way to catch

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 =>

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"

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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 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

Hi, I am facing problem in configuring D-channel. I did the following configuration for TE-120P card /etc/zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 /etc/asterisk/zaptel.conf group=1 signalling=pri_cpe switchtype=euroisdn context=incoming channel=1-15,17-31 DIGIUM card is connected through cable to another end.On placing call from other end to

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

Is there a clean way to disable music on hold for a specific user sip user? I have seen one example that creates a class called [none] that points to an empty directory, which creates log errors that are annoying (but not harmful?) -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is there a list somewhere or a way to find the following: 1- All non US 48 area codes which can be dialed as 1+10 2- All strange area codes which are used for premium services such as 900-XXX-XXXX 3- Anything else that should be restricted if one was to restrict all calls to US 48 only I have found many list but it's tough looking at the entire list of area codes and pulling out each of them

We are looking to be able to put a device in front of an array of Asterisk systems which would do the job of load balancing them. We would store all the particulars on one or more MySQL servers. What want to accomplish is to have all calls sent to/from a single IP, then push the calls off to another Asterisk server in the array. If one server goes out, we are hoping there will be no effect other

I have this in my config: exten => _800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/1${EXTEN}) exten => _1800XXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/tollfree/${EXTEN}) exten => _NXXNXXXXXX,1,Verbose(0,${CHANNEL(peername)} Calling ${EXTEN}) same => n,Dial(SIP/trunk/1${EXTEN}) exten =>

Anyone have any idea if there is some sort of limitation to the number of SIP or IAX end points which can register to an Asterisk system (2.8Ghz dual processor, 2GB ram) while also handling 30-50 simultaneous calls without getting into trouble? Of course the 30-50 simultaneous calls end up being 60-100 channels of mostly G711 VoIP. We have seen issues where our Asterisk just gets all crazy and

Can someone give me a little detail as to what this error message means and why it may be occuring? I keep seeing tons of these roll by on the CLI on one of our systems. Thanks! Apr 9 11:05:40 WARNING[19263]: chan_iax2.c:7538 socket_read: Received mini fra me before first full voice frame

Is it possible to strip the plus sign from the ${EXTEN} var on an incoming call? We have our system setup to deal with incoming calls to numbers without a plus sign, lots of AGIs and databases we don't want to have to change. We have seen things like this ${EXTEN:1} which you can use in the dial command but we want to basically change the ${EXTEN} var right off when it comes into

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.

Hello, I am using 1.4.8 and have a question about Queues. I noticed that if I have an agent logged in using AgentCallBackLogin and that agent is unreachable for some reason (SIP phone unplugged) calls to him/her will completely yack. For example: 1-Agent 500 is the only one logged into queue number 1. 2-A call comes into queue number 1 3-The call is pushed to agent 500 at extension 21 which is

We have issues with all of the SPA-2000 ATAs we have where incoming calls from only one of our Asterisk servers do not complete. Details: 1- On the CLI we see that when the call is pushed to the ATA it shows Busy/Congested 2- We can make calls to this same server just fine 3- We can receive calls from other Asterisk servers running older CVS versions of Asterisk with the same exact ATA

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety

The Asterisk and Zaptel development teams have released Asterisk 1.4.1. This release contains a very large number of bug fixes, including a fix for the recently discovered security vulnerability. It also contains a complete rewrite of the Shared Line Appearance (SLA) support that was first released as part of Asterisk 1.4.0. The new version of this functionality has been tested against a variety