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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

...tal.com-- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > > Dipl.-Ling. Christophorus Laube > > Systemadministrator > > SemanticEdge GmbH > > Kaiserin-Augusta-Allee 10-11 > > 10553 Berlin > > Deutschland > > Tel +49-30-345077-0 > > Fax +49-30-345077-77 > > christophorus.laube at semanticedge.de > > Gesch?ftsf?hrer : Dr.Ralf K?hrbr?ck, Dr. Lupo Pape > > HRB 84682 > &gt...

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

...ers and friends, Im a human living on Earth. ___________________________________________________________ Do You Yahoo!? -- Une adresse @yahoo.fr gratuite et en fran?ais ! Yahoo! Mail : http://fr.mail.yahoo.com -------------- next part -------------- An embedded message was scrubbed... From: duncan@semanticedge.de Subject: ALTGR Problem Date: Sun, 13 Apr 2003 15:41:17 +0200 Size: 2857 Url: http://www.winehq.org/pipermail/wine-users/attachments/20030413/3d86ee01/attachment.mht

Hi there, I want to set up an extension set that acts different depending on the count of used lines. I have a EuroISDN E1 board with mISDN and I only want to offer 10 lines. Therefore I set up a global variables LINES in the general section of extensions.conf and instantiate it with 0. I a call is incoming I check the LINES variable wether is 10 or more. If so I make a call transfer. If not

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, 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 on the list, does anyone of you have experience with asterisk 1.6 and mISDN, pri primarily? Thanks in advance & Regards, Christophorus

Hi, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus

Hi, I installed the current stable wine from the source a few hours ago on my RedHat9 System. Yes, using the --with-nptl switch. Well, most things are running well, but there is a real great problem! When ever I press the AltGr Key, I get the ^ sign and nothin happend concerning the email "at" sign, the special brackets and so on. I have a Toshiba notebook, 2430-101 with a qwertz

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 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