similar to: Distinctive Ring Tones

Has anyone out there had any success with any type of voice recognition? I would LOVE to see something work within Asterisk. I have tried for days and can't seem to get this right. Is it even possible? I have read everything I can find, but it's like the sphinx integration/documentation was dropped because of lack of interest. I tried this with Sphinx-2 because it contained the

Using distinctive ring detection with bell202 cid, is there any way to tell DAHDI to sometimes expect the cid after the 2nd ring, other times after the 1st? I just added RingMaster service (2nd DID w/ distinctive ring) to a TDM800P FXO line. No problem setting dringcontext for the 2nd DID. The 1st DID works normally, but I get no CallerID on the 2nd because the call is picked up before the

In AU we have a number of different dialtones defined for various purposes. >From indications.conf: au <ringcadance> 400,200,400,2000 au dial 413+438 au busy 425/375,0/375 au ring 413+438/400,0/200,413+438/400,0/2000 au congestion 425/375,0/375,420/375,0/375 au callwaiting 425/200,0/200,425/200,0/4400 au

I'd love to get a legacy application running under wine (yeah wouldn't we all I hear ya) and I'm up to a point were it's complaining about a missing DLL as per the subject. I can get the missing file from a Windows installation but no matter where I place it (in the applications directory, in the fake_windows/Windows/System directory it doesn't find it. I have edited the

I have a line from BT (UK) connected to my asterisk system, on a TDM400P. I am able to see either distinctive ring cadences or caller ID but not both. If I try to enable both, all drings show up as 0,0,0. This is a pain because, if I make a call out over that line and the number I call is busy, I can elect to camp on it (ringback), which results in a different cadence of ring when the called

Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you

Surely there has to be one soft phone that works under Linux. I've tried: kphone - it sometimes complains about the need to release the sound device linphone - sssssssslllllllllooooooooowwwwwww iaxcomm - needs some strange widgets various others - either only supplied as binaries, or just plain don't work, or won't compile. Is there just one out there that is guaranteed to work with

Anyone else seeing problems doing pxe boots off the latest dag/rpmforge syslinux packages? Versions 3.35 works, doing the tftp request like this: Jun 18 06:01:19 boothost in.tftpd[25947]: RRQ from 172.16.0.26 filename /centos5-i386/pxelinux.0 Jun 18 06:01:19 boothost in.tftpd[25947]: tftp: client does not accept options Jun 18 06:01:19 boothost in.tftpd[25948]: RRQ from 172.16.0.26

Is it possible to run the Festival command in the same manner as the Background command so that it can be interrupted by caller key presses? -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you choose Linux; when you want a system that just works, you choose

Can anyone _recommend_ a downloadable OSS softphone that _works_ under Linux and is compatible with Asterisk. So far I have tried kphone and linphone and had problems with both, and I am still waiting to hear back from the X-Lite beta folks. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just

Hi, I have been asked to install NUT on a Solaris 9 system. The UPS is an APC Back UPS RS 1500 (link below) with an USB connection. From reading the FAQ, it looks like the proper driver would be newhidups, but there is some development work required to map between the Solaris USB hid and the newhidups. Is that correct, or am I overlooking something simpler?

I am taking a stats class and need to do a single factor anova table with four treatments for an assignment. The instructor showed the class how to do it with minitab, and I am unsure how to proceede in R. Here is the data: I. .8 .2 1.4 .9 .9 1.0 1.0 1.1 1.4 1.2 1.0 II. 1.6 1.1 .2 1.3 1.5 .7 .5 1.3 1.3 .4 III 1.8 .3 1.1 2.0 2.8 1.5 1.3 2.9 1.7 .8 IV. 2.4 2.6 2.82.9 1.7 2.6 2.7 2.1

Okay, I don't have much hair left and I am ready to pull it out. I am running samba 2.2.7a as a PDC for Windows XP clients. Everything works fine for DOMAIN users who are in the administrators group on the XP workstation. My problem is if users are in the Power Users or Users group on the XP client, then when the log out changes to their profiles are not saved back to the server. I know

I'm trying to get the X100P card working in AU. So far I have managed to get it to handle incoming calls from the PSTN and have managed to eliminate pretty much most of the echo. My big problem is getting the outbound calls to work. When I get ZAP to dial out it won't connect and I get what I think is the Congestion signal - like a busy signal but with what appears to be a 10db

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

I recently posted a question RE the Sipura 941 and using different ring tones, Thanks to hads I managed to use SET(_ALERT_INFO=Classic-1) to achieve this but trying this on the GXP 2000's didn't seem to do the trick?? Has anyone one had any luck on this topic? Also haven't been able to find any info on an auto-answer for the GXP 2000, again, I have succeeded in doing so with the

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

When I make a call from a SIP phone to a speaking extension on *, such as one that speaks digits or similar, when I monitor * in very verbose mode I can see it running through the routine associated with the extension, but I am getting no RTP data stream back to the phone. Does the machine housing * need a sound card? Does it need OSS or ALSA modules installed? What actually generates the RTP

Is there anyone using * in AU that has successfully extracted the CLID from an incoming analogue PSTN phone call, and would like to spread the word? Note - I am only interested in analogue, not ISDN phones. -- Howard. LANNet Computing Associates; Your Linux people <http://www.lannetlinux.com> ------------------------------------------ "When you just want a system that works, you

For some thime now I have been trying to get some sense out of Caller ID on PSTN lines in AU and have been getting no where. Now, at last, I seem to be getting something, even though it is an error message (line 5 below). All and any guidance would be welcome. -- Starting simple switch on 'Zap/1-1' Jan 10 09:36:38 NOTICE[18384]: chan_zap.c:5514 ss_thread: Got event 2