similar to: ANI with Pickup application

Hi, How can I integrate Asterisk to Nuance TTS engine instead of Cepstral? Has anybody done this? How is the architecture and can Java AGI be used to communicate between them? regards, Vela Sivasankaran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091021/56254a0e/attachment.htm

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

Hi list, is it possible to get in the running dialplan the status of (SIP) lines without using AGI or anything like that? What I want is a stepwise calling: I have several SIP lines (let's say they are three) which I want to dial to alternatingly. But I do not want to dial to a already busy line and catch the busy. Instead I do not want to dial to that peer but to the next one. I want to have

Hi list, I have bought some Cisco 7941G-GE IP phones and want to use them with asterisk. Before bying I tested the whole setup with three different models of the old 79X0 series (a 7912, 7940 and a 7960). Flashing the formerly provided SCCP-Image to SIP was no problem, but now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but

I have just got a Cisco 7941G and am experiencing the exact same problem (phone is requesting .tlv file from TFTP server and never asks for .cnf.xml file). The phone originally had SCCP on it, but I downloaded and flashed with the latest Cisco SIP image (8.4(3) released 2009-01-13). In reading your message below, it looks like you were going to try an incremental upgrade?did you have any

hi list, I set up a new asterisk machine with asterisk 1.2.13 and misdn 0.3.1rc27. I use an e1 card with sip clients. My extensions look like this: [E1] <snip>...<snip> exten => 33006733,1,Set(CALLED=${EXTEN}) exten => 33006733,2,Dial(SIP/1@192.168.0.23) exten => 33006733-ANSWER,3,Answer() [SIP] exten => _X.,1,Noop() exten =>

Hi list, I am searching for a possibility to do a certain call transfer method which is called "path replacement" in QSIG. But I want to do that in DSS1 (EuroISDN). If my asterisk does a call transfer I want the machine to signalize on dchan that the call path has to be replaced to a direct connect between the caller and the called, i.e. my machine is to hang up after the transfer and

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

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 =>

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

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 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

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

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> | | | | | |

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

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:

OK I am going to do it again. Global Crossing is now sending ANI but it is not in the format I expected. Any one know of a way to get this data into two seperate variables? The first number is ANI and the second is DNIS so it is "*tendigits*tendigits* on one line like below. < Called Number (len=26) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan

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

Is there a setting somewhere in * to define whether I am receiving callerID or true ANI? Global Crossing claims they are sending ANI but I dont think so. My understanding of ANI is that it is always sent, regardless if callerID is blocked. If I dial *67 and my DID, I get "Presentation: Presentation prohibited of network provided number" and no number. Before I call GC on Monday

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.