similar to: Whats required for unblockable ANI / CLID?

I've recently noticed some oddities in my CDR records. In some cases the original CallerID that I've set in the .conf file for the extension showed-up in the CDR as the originating extension (on Zap/ devices on the channel bank), and in other places it was the one that I set using Set(CALLERID(num)=<something>) (SIP/ devices). Digging around a little in the source (and doing some

Can someone in Asterisk'land subscribing to 800 service explain to me how to setup extension.conf to route calls based on the incoming DNIS/ANI. For example I want to route 3 incoming 800# coming across a trunk group to all land in the same queue. So I guess I am asking how to perform DNIS/ANI based call routing? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Due to time constraints, I'm looking to pay someone (by paypal) for a working or/almost-working asterisk skeleton of the examples listed below... SYSTEM INFO: ---------------------------------------------------------------- I will have a single channelized T1 with all lines being available for dialing in, using E&M wink, with ANI2*ANI*DNID being sent to me (as DTMF tones I guess?). I

Hi list I have a requirement that I just don't know how to address - I don't think its strange but can't find any pointers anywhere. I have a user that wishes to have a "multi phone" divert. By that I mean "calls made to his extension say Ext200 can be redirected to a different extension say Ext400 and also to his home landline. Doing the dial is fine using

Here's what I am trying to do... First I'll have a list of 4 digit numbers like: Code:OtherCode 1234:4321 9999:4444 3333:1111 People will call our 800#, Have the number they are calling from read to them (ANI?) & then enter in the code (let's say 1234). If the code matches one on the list, then the OtherCode (4321 for 1234) will be read/spoken to them. With the exception of the

Hi, Returning to my office I find two "missed calls" (from autodialers) that my IP601 displays as originating from 01111111111. However the CDR database recorded the call this way: calldate: 2008-04-04 14:18:16+02 clid: 0172752780 src: 0172752780 dst: 2131 dcontext: default channel: Zap/1-1 dstchannel: SIP/0146472131-007a7e80 lastapp:

I use Trixbox 2.0 with a Sangoma A200 I also have echo so bought the HPEC and yes it works brilliantly. The problem I have is I used to use Trixbox 1.? with this sam hardware and had a few inbound CLID issues on my UK BT lines, Sangoma support suggested changing the RXGAIN in zapata.conf and it worked. Now using Trixbox 2.0 and have upgraged to latest software Asterisk 1.2.16 Zaptel 1.2.15

Hello All, I'm having problems with international calling via Global Crossing. I'm told I need to send a true ani versus a sudo ani. What is the difference and how can I configure asterisk to do this. Global Crossing is denying calls with sudo anis. Thanks, Keith

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Or you could just use "System" and do it the easy way! AGI is better suited for more complicated scripting. In any case, check out the TFOT book - you'll like it. -MC _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Collins Sent: Thursday, February 09, 2006 3:13 PM To: Asterisk Users Mailing List -

Hello, We're having some issues with CallerID and I thought someone here might be able to shed some light as none of our carriers seem to know what I'm talking about. The issues is this: A client of ours uses an after-hours voicemail service as mandated by their corporate office. We have a Day/Night setting that lets them turn this on and off. A call comes in from one of their

I'm running into an issue where subscribers to our service cannot call certain 1-800 numbers if they have a caller id blocked account (restrictcid=yes). This is on Asterisk 1.0.9 and our clients are using Sipura SPA-2002's. Our provider uses a SIP/PSTN gateway, so we hand off SIP to them from Asterisk. The problem appears to be that when an "Anonymous" call goes out, there is

Hi Guys, We have a problem with Caller ID not being displayed. I want to test everything to see where the problem is with the incoming Caller ID and why it's not displaying. I am tracking this down to "Presentation prohibited of network provided number" even though the Caller doesn't use *67 and even though they haven't asked their provider to block their CLID for outbound.

Okay went to: http://wine.dataparty.no/ form there downloaded the rpm, http://wine.dataparty.no/rpm/unstripped/wine-cvs-unstripped-013001-1.i386.rpm downloaded the config file and worked through the install guide. I edited the config file to mirror my window setup: c:\ = \mnt\c d:\ = \temp e:\ = \web f:\ = \documents g:\ = \music h:\ = \mnt\cdrom i:\ = \mnt\zip then as a check did $wine -v and

I receive calls over a T1 with callerid and then *ani*dnis*. I am able to strip out the ani and the dnis in the dialplan but when I try to set the caller ID to be the ani, it looks ok but then if I do a NoOp callerid on the next line, I get unknown. Here is the section of my dialplan: exten => _*NXXNXXXXXX*NXXNXXXXXX*,1,Set(ANI=${EXTEN}) exten =>

Yes its a T3 split into seven trunk groups with one D channel and NFAS on each. Can you explain the cut function or point me somewhere please? Thanks, Steve -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 12:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on

Hi All, My research indicates ANI is not really supported with SIP Channels or passed between SIP servers, even with setting function CALLERID(ANI). So the only place this applies is on PRI interfaces, when sending calls out a ZAP PRI you can set the ANI to whatever and CID Number to a different whatever so on the other end of the PRI you will receive the two different values? Is this correct or

I've got an interesting problem. I just recently upgraded an asterisk server from a may 9th cvs to a 9/29 cvs. When I did the upgrade I was unable to place calls through the PRI. The calls would process fine, but the provider would reject the call and send me a cause 69 or 77 failure code "Unallocated (unassigned) number". Upon contacting the provider, they told me I was

Seri, I think you might just need to find the right variable and pass it to your script. Also, you'll probably need to read up on AGI - I'm assuming that's how you would launch your Perl scripts. Here's the list of variables on the wiki: http://www.voip-info.org/wiki-Asterisk+variables Most likely you're looking for the variable referred to as ${CALLERID(num)}

Good Day list, Need assistance determining the best place to read up on whether Asterisk can help me out. I have a situation where I need to do the following <PRI from Telco> ------- <Analog Channel Bank>------------<Proprietary Box> | | | | | | <PRI Port 1 of Digium Quad T1> <PRI Port 2 of Digium Quad T1> | | | | | |