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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

...X.loads file, I assume. But as I am not getting over this step it stays in the term41.default.loads step, unfortunately. Does that ring a bell to anyone? Does anyone of you have had the same situation? In which state did you get the 7961G? SCCP? And how did you manage to load SIP firmware onto it? Christophorus > I do have to answer to your suggestion of renaming the CTLSEP<mac>.tlv > to SEP<mac>. The phone is still requesting CTLSEP<mac>.tlv and as it > cannot find that it goes into a loop. I also let the phone do that the > whole weekend so there should be no iterat...

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

...llowed_passed_screen) exten => _X.,3,Dial(mISDN/g:E1/${EXTEN},40) exten => _X.-BUSY,4,Busy(1) But whenever a sip client calls to an exten that is busy through e1 I get busy tones for 10s before I get disconnected. But I want to have it only for 1s. Does anyone know how to fix that? regards, Christophorus

...alplan. Using AGI and such things just makes it slower in my opinion (if I call an AGI script that does an "asterisk -rx 'sip show channels' |gawk -F " " {' print $1 '}, for example). Does anyone of you have an idea of how to do that? Thanks in advance. Best regards, Christophorus Laube

...now it complains about a nonexistent CTLSEP<mac>.tlv file. Most of the howtos say something about an empty file but that does not suit to me. Does anyone of you have experience in getting these phones to work or can point me to any information bringing me back in the game? Thanks in advance, Christophorus Laube

...n => 33006712,205,SetGlobalVar(LINES=$[ ${LINES} -1 ]) exten => 33006712,206,Hangup() My problem is that the increment works perfectly, but the decrement is not working. I added the last two extensions only because the hangup-extension did not work. Can anyone of you help me, please. TIA, Christophorus Laube

...e 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 the channels are free again. Is it possible and with what card vendor (mISDN vs.zaptel) and how do I do that? Thanks in advance, Christophorus

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've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm: