similar to: Support for SIP REFER message

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

Hi to all, I'm using a TDM22B. When i establish an external call to the PSTN through an FXO port, I'm not able to know the status of the call (no answer, busy, ...). If I enable call progress (callprogress=yes) in Zapata.conf, I am able to detect the no answer state but if the callee on the PSTN answers the call asterisk doesn't detect that and it jumps to the NOANSWER state and

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 (

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, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera

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

-----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

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

Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code

Is R file IO slow in general or am I missing something? It takes me 5 minutes to do a load(MYFILE) where MYFILE is a 27 MB Rdata file. Is there any way to speed this up? The one idea I have is having R call a C or Perl routine, reading the file in that language, converting the data in to R objects, then sending them back into R. This is more work that I want to do, however, in loading Rdata

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

Anyone know of anybody doing VXML with Asterisk and/or Linux? Tia Kevin

How can I stop the following warning from occuring? invalid factor level, NAs generated in: "[<-.factor"(`*tmp*`, iseq, value = structure(1, .Label = "12", class = "factor")) The Label messages are for "5", "8", "12" and "46". I want the NAs to be generated as needed. Is this causing R to slow down by generating the warning

Firstly can I ask when the documentation site will be online again? I'm struggling here without it. Further to my recent post I have tried to simplify things a little. I have used a VoiceXML app to simple call an asterisk extension. EG: <form id="transfer"> <block> <call name="xfer" dest="sip:101 at 10.0.4.147:5060"/>

Hi If I have the wrong list please feel free to redirect me. I'm running 7.0-RELEASE-p4 and trying to backup to an external USB drive. I'm using the following command dump -a0Lf /backup/diskimages/root /dev/mfid0s1a Where df: Filesystem 1K-blocks Used Avail Capacity Mounted on /dev/mfid0s1a 507630 208436 258584 45% / /dev/da1s1d 709513458

I have a customer looking for an automated way to provide his customers information. He found some windows software called Active Call Center - but I believe that he did the 40 day trial and it crashed his windows machine to much. So he wants something that can do a similar task running on linux. The Users expierience should go something like this One of his customers calls in the IVR asks

Hello, I've seen that Nuance SIP audio provider supports additional information (parameters and extra headers) in SIP URIs, using the format: sip:user:password@host:port;uri-param1;uri-param2?header1&header2 For example, sip:1234@myserver.com;extra_header=Uui?Uui=Hello Does Asterisk support this format? Is there a way to retrieve the value of these additional headers, and then decide

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hi I'm having a problem with ipfw, I think. For some reason it denies packets randomly for example: PING 196.14.239.2 (196.14.239.2): 56 data bytes ping: sendto: Permission denied ping: sendto: Permission denied 64 bytes from 196.14.239.2: icmp_seq=2 ttl=63 time=0.258 ms 64 bytes from 196.14.239.2: icmp_seq=3 ttl=63 time=0.233 ms 64 bytes from 196.14.239.2: icmp_seq=4 ttl=63

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden