similar to: Open Source QoS .

Buenos días; Estoy intentando aprender algo de sobre los modelos lineales mixtos. Estoy siguiendo el libro "Multilevlel analysis: techniques and applications" de Joop Hox. Estoy intentado reproducir las tablas y gráficos que trae el libro con el paquete lmer. En este libro se indica que en la comprobación de los supuestos del modelo hay que estudiar los gráficos de residuales tanto

Hi All: I'm a newbee to Asterisk. I currently working on a project and want to know if Asterisk does support R2 Signaling. Thanks Begra8fl >From: asterisk-users-request@lists.digium.com >Reply-To: asterisk-users@lists.digium.com >To: asterisk-users@lists.digium.com >Subject: Asterisk-Users digest, Vol 1 #3647 - 9 msgs >Date: Tue, 04 May 2004 13:32:00 -0500 > >Send

Gautham Kasinath Software Engineer Arkin Systems Pvt Ltd T. Nagar Chennai Ph. (91) (44) 8216686 Extn 14

I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/ba256d25/attachment.htm

Hi, I am thinking of signing up with voice pulse connect to connect to my asterisk server and using it as a regular line. Is it good? Or should I go with vonage or others ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/16792f10/attachment.htm

I am thinking of dumping broadvoice so I need some other VoIP providers that have a las vegas DID and a service better than broadvoice. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050131/bf4b1c31/attachment.htm

I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden

I finally got my incoming and outgoing to work on Broadvoice with a configuration file that is no where close to the one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com

Is the Broadvoice service up? I just signed up with them and started receiving calls in no time but could not make calls. And after a few minutes I cannot even place calls. register => [number]:[password]@sip.broadvoice.com [broadvoice] type=peer fromuser=[number] host=proxy.lax.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband any help would be

Hi, Ive just patched Solaris 10 which has for some reason broken the samba config The csw version of samba is being used (not sure if that will make any difference) but after the patching the version changed (details below) and now no-one can map their userid. Basically, when someone tries to map through Windows explorer they get prompted for a userid/password. After entering the Unix credentials

Hi, We are looking towards a 8+ CO line setup (20 extensions) in our office but we do not want an IVR(auto-attendant) feature. All incoming will be answered by a receptionist. I have read the multi-line configuration for cisco 7960 thread in this list but that way I believe we could only display 6 incoming lines. What will happen to the rest? Does the expansion module for the cisco 7960 work

Which would you recommend as far and quality and pricing to connect to asterisk (including DTMF issues)/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050125/bf7bf081/attachment.htm

Is there any way to change the outgoing caller id on BroadVoice I have tried SetCallerID(Name <Number>) but that does not work Thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050126/bbd95fcf/attachment.htm

BDC to a SAMBA PDC. Did I read this correctly? Has anything changed since then. If it can't be done, can anyone recommend another solution that has worked for them? I also would like it if SAMBA, Courier, Postfix could share the same username/password file. All of the above I mentioned (According to my reading) support the reading of /etc/passwd, so why would one want to use PAM or LDAP

I have been looking around for Outlook Integration for Asterisk. Saw the Asterisk TAPI wiki page and also ran across this: http://www.fonality.com/pop.cgi?page=pop_pbxtray.tt (PBXtray) It looks like Fonality has managed to make an app that does screen pops and allows click to dial. Has anyone else been able to get this all to work successfully? Looks pretty slick.

Just got a headset for testing asterisk and am using X-Lite. I plugged in the headset into the headset jack and is there any way to configure X-lite to use the headset instead of the speakers? Or will I have to plug the headset in the speaker jack ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

using Asterisk. =20 Is this all you want Asterisk to do? (eg as an application service rather than provide telephony for the rest of the library as well), or are you looking to have it replace your existing telephony equipment? =20 As a suggestion if you google Trixbox and Nerd Vittles you will find a fairly detailed explanation of how to set your Trixbox server (a version of Asterisk) up to

I did a make webvmail and I get the following error on redhat 9.0 No HTTP directory make: *** [webvmail] Error 1 I have the perl-suidperl rpm installed and apache installed Thanx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050501/275c1e94/attachment.htm

Hi, I was going to order the T100P but it is replaced by TE110P. On further reading the TE110P does not need an external router (The one that separates the data from pstn lines ?). Has anyone got it configured? And on the wiki it says that the drivers for some distros don't exist yet. Is redhat supported? And if I need to connect a fax machine will the FXO cards on ebay work (I know to

An example for Radius is calling cards.. I can use * for this kind of service... With platforms that use Radius Server. -----Mensaje original----- De: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Lunes, 17 de Noviembre de 2003 07:16 p.m. Para: asterisk-users@lists.digium.com Asunto: