similar to: Syntax for hiding caller ID but still passing ANI?

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. I need to send calls to different programs depending on where the call originates. For example, I need calls from San Diego (NPA 619 and 858) to to be routed differently than L.A. calls. I tried entries like: exten => 4044633/_619.,1,OurApp,sandiego-queue exten => 4044633/_858.,1,OurApp,sandiego-queue exten => 4044633/_213.,1,OurApp,losangeles-queue exten =>

For our application we want to read the callers number back to them (calling in on a 800# that would somehow point to multiple lines -- figuring that all out still). I was under the impression that calls to 800#'s are unblockable by the caller (irregardless if they have caller id blocked or not). I'm not sure if this is because of ANI or CLID (or if ANI _is_ CLID) or how any of this

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:

Quickie: Does anyone out there have experience with PRI delivery of ANI II information? Specifically, I want to know if it's possible from within Asterisk to know if the inbound call (which may or may not be to an 800 number) came from a payphone or not. I know with some 800 providers it's possible to block inbound calls from payphones (due to the FCC surcharge etc) but was wondering how

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

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

With the help of one of the providers we terminate on, I've found the source of the problem of getting busy even when the called isn't really busy in the absence of ANI codes in sip headers generated by asterisk. If I put a NoOp(${CALLINGANI2}) in the dialplan before the dial I can see it holds the value '0', but seems that value won't find the way to the sip header. Is

We are using TDM40B's to connect some devices to Asterisk which depend on caller information arriving as ANI, rather than as Caller ID. I am unsure if the TDM40B supports this in the first place, and if so, I am unsure how to configure it so. I've searched the wiki but couldn't find anything. Can someone please confirm whether or not this is possible? As a fallback I could reconfigure

Alexander, Thanks a million, you really saved me some time on this one. -----Original Message----- From: Alexander Lopez [mailto:Alex.Lopez@OpSys.com] Sent: Mon 4/10/2006 1:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: RE: [Asterisk-Users] ANI and DNIS Seperation on a PRI(TelephonyNumbering Plan (E.164/E.163) (1)'*4105556654*8005550215*' ]) from the

I set up the Asterisk for my company which is a business center, I will assign a specific telephone number to my client that uses my serivces. All of their incoming calls will be first picked up by the receiptionist, can I disply the company name instead of the called number on my receptionist's telephone display, so that she can answer the call with the right identity at once... Regards, ML

does anyone know how to route a call coming in with ANI*DNIS* Oct 31 12:47:41 VERBOSE[30056] logger.c: -- Executing Set("Zap/49-1", "DID=1231234*4812*") in new stack I tried making a route for _.*4812* but that matched everything rather then just the dnis i wanted.. any ideas? I would preferably like pass the callerid along to my extensions, but for now the important

Hi, does anyone of you have made it to get the ANI also picked up? I mean: if I fetch a foreign call to me by using the pickup application I want to see the callerID/ANI of the caller to the foreign extension. Is that possible and if yes - how do I achieve that? Regards, Christophorus

When Asterisk sends a call to "a phone company" via a PRI/Dahdi, does it actually send CALLERID(ANI), or only CALLERID(NUM)? Cary Fitch

Hi folks, I know this isn't an Asterisk question, but I'm really desperate and wondering if someone could help me. I apologise for the off-topic post. Cisco phones connected to CallManager can forward calls. But when they do, CallManager conserves the originating caller's ANI in the new leg that is built. I cannot find a way to get it to rewrite the ANI to be that of the phone.

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

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

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

Hello, Being totally fed up with the lack of quality and reliability from both VoicePulse and BroadVoice, We are switching to a direct IP connection to Global Crossing. We've installed a local point-to-point T1 into their CO, and they will give/take SIP g729a directly and act as the gateway for us. In setting up the inbound SIP service, they are asking the question, "In what format do

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