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...he website. Only 1.95 is available. Will that work? Does it need the patch mentioned on the Wiki page? Does anyone have an apt compatible repository of everything needed to get Festival working with Asterisk under Debian Woody? Thanks in advance Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: darryl@oeg.com.au Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole "There is more than one way to do it" slogan, you have to give someone a swiss army chainsaw ...

...-- 1 darryl users 23411 Aug 19 16:28 utt.feats I've tried setting a number of different voices. Some don't give any audio and some only seem to give a small 'blip' of sound. Does anyone have any ideas? Thanks in advance Darryl -- Darryl Ross Senior Network Engineer OEG Australia Email: darryl@oeg.com.au Phone: 08 81228363 Office: 08 81226361 If you want to live up to the whole "There is more than one way to do it" slogan, you have to give someone a swiss army chainsaw ...

Dear all, I have an Asterisk box serving as gateway to a set of POTS phones that all share the same telephone number, and I register this box as gateway to a H.323 gatekeeper with the telephone number as the gateway prefix. I am wondering if there is a way to have Asterisk-wide (perhaps to be called 'universal') variables that can be used in all config files. As far as I know, now e.g.

Has anyone had any luck with the DEV LITE KIT? I'm getting very erradic behavior. I'll start it up and it will answer maybe one or two calls then hang leaving my phone in an off hook state or giving a loud shrill tone. It keeps saying "Red Alarm". Once it hangs it will not answer the phone again until I reboot. Starting and stopping Asterisk or reloading the drivers does not

Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and

...user, group, acl from NT on the Unix server. If it is possible, where i can get the rpm?s to install it on the Red Hat Unix 6.2 ????? Thanks Thomas _____________________________________________ Thomas B?hler tboehler@b-c.cc b&c Software Entwicklung OEG http://www.b-c.cc Tel.: +43 5574 62895 Fax: +43 5574 62895 4 Mobil: +43 664 3410046 -------------- next part -------------- HTML attachment scrubbed and removed

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

"In general you can match callerID with the /, but if you don't put anything after the /, then the rule matches "no caller*ID", and if no slash is there at all, it matches "any callerid". " Ok.My question is -> how to match callerid from 001... ? and if don't know how many numbers ? exten => s/0_,Answer don't work- anything else ? tnx Thomas

Hello list. I'm trying to rebuild the 2.6.9.67.0.20.EL kernel, but it fails even without modifications. How did I try it? Created a (non-root) build environment (not a mock ) Installed the kernel.scr.rpm and did a rpmbuild -ba --target=`uname -m` kernel-2.6.spec 2> prep-err.log | tee prep-out.log The build failed at the end: Processing files: kernel-xenU-devel-2.6.9-67.0.20.EL Checking

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Hey All, Having a bit of a problem with a Wildcard X100P card. When I try to make an outbound call using the card, it picks the line up and then only dials a single digit. I've confirmed it's only dialling a single digit by listening on a phone plugged into a parallel socket. Incoming calls work fine, it's only outbound calls I'm having a problem with. I've tried

Hey All, We are running a small SIP/IAX termination service at the moment (planning on growing it) with 2 asterisk machines. One terminates the SIP/IAX calls from our customers and one is our gateway to our upstream provider. Both machines are logging CDR data to the same postgres table using the cdr_psql module. The problem I am having is I'm trying to work out how to link the CDR records