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...t formats, but only ogg is supported. Is that right? Is there a place in the decoding chain, where the information is in a frequency domain, and where one could more or less easily change pitch by shuffling FFT (or alike) coefficients around? Thank you for your help! With best regards Hartmut Niemann --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages...

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

...ted. From the code it looks like being intentional. In my opinion the documented behaviour makes more sense, but that's up to you, I would run it from a shell script anyway ... As I am not on the mailing list, please CC replies to me. Thank you for your attention! With best regards Hartmut Niemann --- >8 ---- List archives: http://www.xiph.org/archives/ Ogg project homepage: http://www.xiph.org/ogg/ To unsubscribe from this list, send a message to 'vorbis-dev-request@xiph.org' containing only the word 'unsubscribe' in the body. No subject is needed. Unsubscribe messages...

Would it be possible to execute some kind of script when for example Asterisk restarts... or stops... ? How can one read the status of Asterisk so that when the service is stopped I could be notified by mail, by text message,... ? I don't know how to read the status of Asterisk (or the change of status) in a bash-script. Thanks for the reply ! Kind regards, Jonas. -------------- next part

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic

...ifferentiate between manager skill and luck. The second resamples portfolios to produce alternative asset allocations and to provide rules for rebalancing. In each case S-PLUS, coupled with S+NuOpt are in a unique position to provide quick computational solutions to these problems. 4:00 PM Douglas Niemann, Zurich Financial Services Abstract to follow 4:45 PM Closing Remarks 5:00 PM Cocktail Reception Register Now! http://www.insightful.com/news_events/UKfinance05/default.asp There are limited spaces for the event, so please register ASAP. If additional places are needed please ensure colleag...

Hello, Let me explain a scenario There are different Asterisk Servers at different Remote locations. Recording in different formats for FIVE seconds reveals that Format : Size wav : 84 KB gsm : 8.3 KB sln : 84 KB It can be recorded in any format. This is size for five seconds only. We need to transfer these files from different remote servers to a centralized server. We need to play these

Hello, i have a problem with the context parameter in the sip.conf. i'm using a german sip provider (sipgate.de) and everything worked fine in asterisk 1.4, but on 1.6.1 i got the following error message: NOTICE[3071]: chan_sip.c:18160 handle_request_invite: Call from '' to extension '8001187e0' rejected because extension not found. sip.conf: register =>

Hi, My customer has a outdated firewall that is also presenting a NAT nightmare for getting the Asterisk server reachable from the internet. What firewalls work good with VOIP? I really want to steer away from any ALG supported firewall. I just want a good firewall that works well with Asterisk. Thanks, David Wathen -------------- next part -------------- An HTML attachment was scrubbed...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm trying to connect to ekiga.net through a client connected to my Asterisk server. For it I am being based on this [1] document. Next I put the configurations that I am using. /etc/asterisk/sip.conf: ; Outgoing to ekiga.net [ekiga] type=friend username=MyUser secret=MyPass host=ekiga.net canreinvite=no qualify=300 nat = yes stunaddr =