similar to: Voice XML Asterisk Integration

A couple of weekends ago I decided to see if I could get Asterisk to play nice with TellMe's VoiceXML studio. They provide the VoiceXML studio for free, and you can access it through SIP, so I thought this would be a fun and cheap way to integrate voice recognition into my IVR. I have posted a brief tutorial with code and examples on the voip-info.org wiki (

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in continued evolution of

Hi All, Do any of you know what the status is for VoiceXML support in * ? Is it already existing, or is it planned for the future? If it's not in now, do you know on what type of scale the work would be to integrate VXML into * ? Thanks in advance

Hi Freinds, I am trying to do the following. 1. Accept the call from call ifle. 2, Answer it 3. Extract the dial number and variables from the call file request. 4. Pass that parameters to voiceglue 5. Catch the parameters (dialnumber and cli) in voiceglue.conf 6. Point to the voice.xml file dynamically by matching number I want to make the line in voiceglue.conf as <DEST_NUMBER>

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to upload woth TFTP due to some reason it's getting failed. Do I need to load 3pcc firmware or anyway to Configure from the phone itself or from the GUI? I have the SEPMAC.cnf.xml as well. Any suggestions would be appreciated. Regards .

Can't I upload the 3PCC firmware ? available from the Cisco website? Actually it came with sip88xx.... firmware. Regards . On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies147 at gmail.com> wrote: > Hi, > > You have to buy the 3PCC version for this to work. Once you have this, > they work very much like the Cisco SPA handsets. > > I also ended up with a non-3PCC

Actually now I have the phones with SIP firmware. I will try with 3pcc firmware along with XML files. Or any idea if we have CUCM application can we change the firmware. am ready to buy the developer edition. Regards . On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies147 at gmail.com> wrote: > I tried... repeatedly... I failed. I bought some 3PCC phones, and they > just worked.

Hi All, i am stuck in NAT issue on ec2 cloud computing from last 2-3 days , may be some of you are doing setup and integration on cloud. below is my setup details which may help you to suggest me solution. Asterisk version : 1.6.2.6 1) Kamailio server having public_ip as well local ip .i am using mediaproxy [also tried rtpproxy] . 2) Asterisk server having public_ip as well local ip. setup:

Hi I need to know if Asterisk supports the full features of the SIP REFER message (i.e blind and supervised transfers). I'm trying to do a supervised transfer through Asterisk from a VoiceXML application using the <transfer> tag and setting bridge="true" (i.e <transfer name="transfer1" bridge="true" connecttimeout="10s"> ) but as soon as

All; Since I have interest in providing the capability for callers to speak the department, person or number they wish to call, as well as other IVR scenarios, I have been reviewing much of this lists email archives and searching the web for open source voice recognition that will work with the Asterisk PBX. What I am trying to determine, is what will it take to get it working on Asterisk? How

What voice recognition software options are available with Asterisk? I've seen some references to Sphinx in the lists and docs, but I can't find details on on how to integrate with Asterisk. If anyone has had success integrating and using voice recognition in Asterisk, I would greatly appreciate advice and links to information. Thanks. -- David -------------- next part --------------

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Milan! Thanks for your comments ont he requirement list. [Milan Zamazal, Dienstag, 26. Oktober 2004 20:58] > [Since the mailing list apparently hasn't been created yet, I continue > in private not to freeze the discussion for too long.] > I have just asked David Stone when we can start using the list. > BTW, this might be

Hi, I am working on vxml to record voice. I have trouble with getting url of recorded voice. I want to save and I am using java to get record parameter from url and it returns a string which is audio/basic:len(123123):p0x5a6e6241, but I want to get file object or base64 string with parameter or to relate returning string with path in asterisk server, are there any way to do this? --

Hi, I created a extension in Asterisk, the extension has been configured in Android softphone 3cx. When I tried to call from Andorid phone to some other IP extension which is registered in Asterisk, I am not able to hear the voice, when I check the asterisk log or wireshark there is only one way RTP traffic, from Android I am connecting to Asterisk via 2G GSM network. Any idea would be

Anyone here technical enough to design a voice recognition voice xml interchange for asterisk please email me; I've been speaking with a contact of mine that is in the voice recognition space and he is interested in 'donating' some technical support to the Asterisk community to assist with this project. This can only help benefit the Asterisk Community if this comes off. If

True agree, problem is somehow the people purchased.... am supporting to overcome that. Trying level best... around 20 phones has been purchased.... ?? On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigoseth at gmail.com> wrote: > With all the money you plan to invest in firmware, licenses, etc., you > have bought a Grandstream IP phone or Yealink... > -- >

Hi to all, I am sending a SIP REFER message to Asterisk from a VoiceXML application using the <Transfer> element to do a Transfer through Asterisk. I need to know if Asterisk supports the full features of the SIP REFER message because if i set 'bridge=true' in the <transfer> element of the VoiceXML application to supervise the call, Asterisk sends a NOTIFY message with

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

Hi, As Freelance programmer/consultant I'm looking for project/task of IVR/CTI/CRM/IP-based, my skils are as following, 1 Dialogic-based CTI/IVR software programming 2 Intervoice IVR development 3 Siebel CRM integration and development 4 IBM DirectTalk and WebShpere Voice(VoiceXML,...) 5 IP-based development(VoIP,h323,sip,...) 6 Cisco

Hi. I am new to asterisk. I have an ivr application with asterisk and voiceglue. I make a call from asterisk (say to A) and when callee press a button voiceglue transfer the callee to another number (say to B). When I look cdr records the billsec between A and B always 0 and billsec with A shows the billsec for A and B. I am confused. Is there a reliable way to get the real call durations? Best