similar to: Asterisk - VoiceGenie IVR

Hello, Is there anyone who has ever done a setup of VoiceXML combined with some licenses from Nuance for the ASR/TTS engine within Asterisk ? I'm currently working with VoiceGenie, for the VoiceXML + ASR/TTS engine, but we are having a couple of issues which I guess are caused by VoiceGenie. If there's an alternative, it would be very interesting for us. Thanks, -- Eric Rousse

Hi, I was planning on getting a Dell PowerEdge 2950 for our new Asterisk configuration. But while searching for documentation about it and/or reported issues, I found this: http://www.voip-info.org/wiki/view/Asterisk+hardware WARNING - many Dell motherboards use the e1000 gigabit ethernet chipset, which has been known to cause random locksup - if you plan on using a Dell server, disable the

Hello, I am running a samba 2.2.7 server that currently belongs to a NT4 domain (security = domain and password server = <pdc name>). There is a plan to roll out a Windows 2003 server with it's own active directory domain. Can my current samba server allow users from both domains to mount its shares? If so how? Thanks, -- Neil Watson Network Administrator www.voicegenie.com

I am experiencing a "606 not Acceptable" error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried

Hello. I'm trying to upgrade my asterisk installation to most current CVS version. Currently I am running CVS-03/24/04-07:26:16 and dialing out works fine. When I install the latest CVS, outbound dialing fails, but inbound and internal calls work just fine. == Spawn extension (it, 9651246****, 2) exited non-zero on 'SIP/8202-d359' -- Executing

Hi, I have a basic dial plan question; Here is the scenario. Call comes through IAX and my * authenticate, then collect the digits and dials out, simple :). Here is the dial plan; [did-in] ;for did callers exten => 866219xxxx,1,Ringing exten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten =>

I just tried to set up a tinc test installation, but I guess I did something wrong, because I can only ping one direction. Here's my setup (lan1,192.168.99.0)<-->firewall1<-vpn->firewall2<-->(lan2,192.168.100.0) firewall1 has tap0 on 192.168.88.2 firewall2 has tap0 on 192.168.88.3 i set up the following routes : on firewall1 : 192.168.100.0 gw firewall2(192.168.88.3) on

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

Hello list, somebody can helpe, my problem is when I run this command ./samba-tool dbcheck Checking 6205 objects ERROR: missing backlink attribute 'memberOf' in CN=Sandy,OU=Administrador de Red,OU=Comercializadora,OU=CUPET,DC=eccmg,DC=cupet,DC=cu for link member in CN=SHARE\0ADEL:ddacaf41-c14f-4e7c-8606-704eec753a4f,CN=Deleted Objects,DC=eccmg,DC=cupet,DC=cu Not fixing missing backlink

The situation is that I want to have certain extensions that can only call nationwide and others that can make international calls (with a different provider). I suppose, you could also see this as a scenario that could involve multiple internal extensions that get matched to specific outbound trunks. I have tried to find appropriate commands in the Asterisk documentation, but could find nothing

Hi, I'm running samba-2.0.4b rpm package on Redhat 6.0. I was able to setup samba just fine along with the printer. I can print from the linux box just fine both text and postscript. My NT 4.0 server(SP3) running PDC saw the printer fine. I was able to install the printer as a network printer with no problem. But I just cannot print to it. When I turned on the debug level to 10. I got the

please help!!! I am dialing into our asterisk server(TDM400p) from the psnt. I hear our voicemail message and I press the extention 1000. The Polycom ip phone in the office rings. I pickup but neither side can hear one another. What have I done wrong? thanks sip.conf: [general] context=local-access ; Default context for incoming calls bindport=5060

Hi List, Any idea how to get wireless lan : 03:00.0 Network controller: Intel Corporation Centrino Advanced-N 6205 (rev 34) working on centos 5? Eero

Hi all I've only been working with Asterisk for a matter of days but have already grown into a big fan =) Much as I've managed to get internal calling working fine, I have a configuration running on an old PII-233 on RH9 with a (although not badged as is a) Dynalink IS64PH/Winbond W6692 PCI Card as /dev/ttyI0. The card works fine in minitel and dials out without a problem.. However try

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to

I'm a newbie with a TDM11B. I've read the FAQs about linking /usr/src/linux-2.6 to /usr/src/linux-2.6.8-1.521 and /lib/linux-2.6 to /lib/linux-2.6.8-1.521 but still get a million errors and eventual abort during compile. Could someone point me in the right direction? I do a yum update every day and I'm using the CVS from 9/14/04. I'm also using an ASUS CUSL2 system board.

Jeremy, At this point the belief is that there are no issues left blocking removing autoconf. The plan is to remove it after the 3.8 branch. In case you missed the thread where that was decided it is here (http://lists.llvm.org/pipermail/llvm-dev/2015-November/092150.html <http://lists.llvm.org/pipermail/llvm-dev/2015-November/092150.html>). This discussion has been going on for over a

================================================================ "You can''t have everything. Where would you put it? Steven Wright ================================================================ Release Announcements ===================== This is a maintenance release of the Samba 3.2 series. Major enhancements in 3.2.11 include: o Fix domain logins for WinXP

================================================================ "You can''t have everything. Where would you put it? Steven Wright ================================================================ Release Announcements ===================== This is a maintenance release of the Samba 3.2 series. Major enhancements in 3.2.11 include: o Fix domain logins for WinXP

I have to design a dialplan for mulitple contexts (multiple companies) and I'm not sure how to go about it and I thought someone may offer help. Here is some background. There are three separate companies, let's say A, B and C. Each has their own context and each has their own set of numbers (these are just examples, not the actual config): [ContextA] exten =>