similar to: Re: [Asterisk] phones being autoanswered?

Is it possible to give a false ringbakc on asterisk ? -- Michael Bielicki Managing Director TAAN Consultants Ltd http://www.global-gateway.net/ -------------------------------------------------------------------------- This correspondence is for the named person's use only. It may contain confidential or legally privileged information or both. No confidentiality or privilege is waived or

Hi, I cannot get SIP channel working with folowing codec configuration: [sip] disallow=all allow=g723.1 ;I need this codec between sip phones (BT100) allow=ilbc ;Use this codec to others Calling between BT100 SIP phones is OK - asterisk makes native bridge (with g723.1) between them. When I'm calling from SIP to other channel (iax,zap,...), asterisk is not able to chose right codec

Dear colleague, I'd like to compute linear discriminant analysis, using R. In the book Modern applied statistic with Splus (Venables & Ripley, p. 396), lda function is used. Could you tell me where I can find this function? At what site, can I download this library ? Thank for your help. Best Regards Sovan ---------------------------------------------------------------- Prof. Sovan

This was published on O'Reilly's site last week, but they didn't tell me until now. :) The article is pretty minimal, because I had a limited number of words to work with, so many features are not implemented. However, it's a good start. http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html I had a few people review it beforehand, but I'd also welcome changes or

Hi there, bristuff comes with these two applications - and too little info to understand what they are for. Anyone has a clue and is willing to share it? Thanks, Philipp -= Info about application 'Autoanswer' =- [Synopsis]: Autoanswer a call [Description]: Autoanswer(exten):Used to autoanswer a call for an extension. -= Info about application 'AutoanswerLogin' =-

Hi. I was wondering if the zplex in the dev kit could be configured to have all fxs ports, instead of the standard 8 fxo + 16 fxs. If so, anyone managed to do that? Matteo. -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -----BEGIN GEEK CODE BLOCK----- Version: 3.12 GS d? s:- a- C+++ UL++++ P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e

Hi 2 all. I've a nice issue my * system. I have: asterisk latest cvs T100P + Zhone cb (3fxo+10fxs used) Diva Server 4BRI running Capi a bunch of pots and sip phones The whole system is running good (thanks to capigod and Bic. for the help with the Eicon Diva), but there's a very strange issue when calls arrive on the Diva and then bridged to the pots on the channel bank: the voice is not

On Wed, Feb 01, 2006 at 05:28:40PM -0800, Peter Michalek wrote: > The patch you suggested works OK, I tried it on the snapshot of 1/28/06 > using a user authenticated via GSSAPI/Kerberos, with this result, which > I think is acceptable: [...] It's not clear to me from the output, but does the connection close after the PAM account check failed? > Could we make this part of the

Is it intentional that password auth using PAM continues trying to log on (giving password 3 prompts) in the case that a user is disabled (so that pam_account returns an error code). It can be argued both ways (saying 'you are disabled' is giving out too much information, making it look like you are entering the wrong password confuses and frustrates the user)

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

It was one year ago today that I released a patch to get OpenSSH compiling on Linux. I had no idea just how much trouble releasing that patch would get me into :) Within days I was inundated with patches, improvements and portability enhancements - contributions which have made portable OpenSSH the success it is today. So allow me to thank the current developers and all of you who have

Security=server Username map=/etc/opt/samba/username.map All NT ids are mapped to the same unix id via username.map. Some NT id's don't see all the SHARES when they access SAMBA...Not sure why this would be if all NT ids are being translated to the same unix id.

I'm replacing a Merlin for a client and they have a PagePal Intercom that I would like to reuse. Here is what I know about it: It has a screw-down wires that goto rj-11 (This was told to me over the phone) that went into one of the Merlin ports. I tried bring it up with fxo_ks and fxo_ls (assuming it was analog and autoanswered) but no luck. I would be happy to replace if anyone knows of

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

I'm looking for anyone who has some vendors for rackmount PC hardware in a fairly decent configuration for Asterisk/Digium T1 termination. I am looking for well-built hardware from an integrator that understands correct component choice. Preferably, I'd like SCSI but I might be able to do Ultra-IDE platforms. I need at least two PCI slots in the chassis (so 2u/3u size factor is OK.)

Hi, Has anybody used AVM C1/C4s on Asterisk? What's the result? I know there may be better cards, but I have 2 C4s which I want to put to good use. Andrea Coppini +356 79 ANDREA (263732) andreacoppini@iwg.info EMPOWER PEOPLE - THE WORLD IN YOUR HAND iWG (iWORLD GROUP) is a global e-mobile company creating, building and growing new businesses. iWG founders are pioneers in creating

Hello. I have just arrived to Asterisk. I would like to know if Asterisk can perform some functionalities I am looking for. I want to allow voip over sip to some users. All of them must have their own user name and password to login to Asterisk so only allowed users can login. All calls started by users have to be redirected to one account at our voip provider. I think those functionalities can

Can someone point me in a direction where I can acquire the Cisco 7960 SIP firmware? I just purchased two phones of ebay, but they are both in Call Manager mode. Thanks, Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030515/6265b777/attachment.htm

I just switched my rails app, which I had previously been running on my pc, to my linux server and have decided to use mongrel. However, since doing so I get an error when I first access the app saying "no such file to load -- irb". After I refresh this error goes away, but I keep getting the following errors in my logs, and I''ve noticed that certains things, like sessions and

Bonjour, J'ai un petit souci avec mon petit couple rsync over ssh. Quand je fais un rsync d'une machine, et que celle-ci dispose d'une messagerie de type vpopmail, le backup m'indique ? la fin que certains fichiers n'ont pu ?tre transf?r?s. Ceci est du au fait que sachant que les services POP/SMTP sont actif, des fichiers ont ?t? supprim?s entre le temps ou la liste des