similar to: behaviour depending on count of used lines

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, 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

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

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

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

Dear all, I use Dovecot SASL (2.1.15) on Ubuntu 12.04 for IMAP authentication and Postfix SASL authentication. I tried to setup SCRAM-SHA-1 as SASL mechanism. This works well on Dovecot's client side towards my OpenLDAP server (with libsasl-2), but fails on the server side (IMAP and SMTP). In the following, there's an extract from Dovecot's log, when using mutt as SMTP client:

I am porting over the portable version of openssh to our uCLinux implementation. Everything has worked with minimal effort and I appreciate all the work. But, I am having a problem whereby the sshd executable is crashing and I really could use some help on where to look at this in more details. Here is how I start up the sshd for testing. /usr/sbin/sshd -D -ddd -f /etc/ssh/sshd_config -p 65

Hi all, 1. We are using Linux kernel 2.6.20.11 64-bit and Samba 3.0.28a 2. We are trying to connect from this linux machine to a Windows ADS running on a separate Windows 2003 system (WINADS machine). 3. Though we are able to retrieve the name of the WINADS machine from linux (We see the name of the machine in samba log file), we are unable to access any of the users in the WINADS

Hello people: I''m new in the forum. I''ve implemented the script for load balancing of "Linux Advanced Routing & Traffic Control HowTo" and I''ve a question: When I run the next command : "ip route" I get the folowing information: -------------- 192.168.0.32/27 dev eth0 proto kernel scope link src 192.168.0.33 192.168.0.96/27 dev eth2 proto

hi i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm:

Dear all, Does R offer a means by which a function can determine whether its return value is assigned? I am using R 2.4.1 for Windows. Suppose what I am looking for is called "return.value.assigned". Then one might use it like this myfunction <- function () { # Create bigobject here if (return.value.assigned()) { bigobject } else {

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

Hello. I set up my samba server and added two users, "alvarezp" and "lunablack". The first one works, but the second one gives Access Denied errors. I already tried to map "lunablack" to "luna" via 'username map', with no luck. Both passwords are under or equal 8 chars long. Here is some info: root@alvarezp:/home/alvarezp# cat

Hello, this is an example extensions.conf. [default] exten => 500,1,Answer exten => 8,1,SetGlobalVar(firstdigit=8) exten => 8,2,Goto(process,s,1) exten => 9,1,SetGlobalVar(firstdigit=9) exten => 9,2,Goto(process,s,1) I call extension 500 and send dtmf digit 9. This is printed to the CLI: -- Executing Answer("Zap/20-1", "") in new stack -- Accepting

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

How do you enable call waiting on sip phones? Ive looked and googled and can only find call waiting pstn phones butnot for sip. Is their a way of setting this up within the dailplan?

I'm still trying to learn some parts of Asterisk, so sorry in advance for the dumb question! How do I set up an extension to dial out to the PSTN through my ZAP interfaces? I want the ability to have a ring group that will ring all of the phones in an office and then ring cell phones if nobody answers. I'm sure this is simple to do but I'm at a loss. I have tried the following