similar to: Help with follow me

Hello all, Has anyone done one of these? I''ve done similar things in java and php, by having easy access to SQL statements. I am new to Ruby/Rails( I''ve been at it for 10 days ) and thus far find it extremely confusing. I''ve got a bit more of a handle on it from the Agile book that someone loaned me, but I still find it hard to understand. Anyway, I''ve got

I am having trouble building appradius from http://appradius.minitelecom.org/ I configure, make, make install cpprad-1.0, but when I configure, then make appradius I get :- obelix:/usr/src/appradius/appradius1.0 # make make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib' make[1]: Nothing to be done for `all'. make[1]: Leaving directory

My asterisk server is in the cloud. Figuring out how to send an email is too much brain damage. So i can't use the email feature that's built into voicemail. What I want to do is execute a remote command with the voicemail as an argument. The remote machine command would email the message. I'm thinking of: same =>n,VoiceMail(vm,u) same =>n,System(ssh myserver "emailVM

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > George Pajari > Sent: Thursday, June 09, 2005 10:19 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Voicemail and MS Exchange Synchronization > > > We have a customer considering

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

Good Evening, I was wondering if anyone knows how to clear a share on a server using a command line interface ? Basically I have a [public] share that I would like cleared down every night, I've worked out that smbclient //apps-srv/public -N -c "del *" > /var/log/samba_public_delete.log 2>&1 seems to go most of the way to doing what I want - however it doesn't

function <- function(p){ i <- 2 r <- 0 while(i < p) { q <- (p/i) while(q>=1) { q=(q-1) } if (q==0) { r=1 } else i=i+1 } return(r) } This function stays in some kind of loop when I enter 4 as a function value. It immediately returns 0 when I inter 3 though. I don't seem te get why it does not give me just a result. Can someone tell me why it stays in a loop?

Tom - here are a few suggestions Plantronics 510SL Bluetooth - they make a bundle that comes with a Bluetooth transceiver and a handset lifter. This is nice if you have users that already have Bluetooth headsets for their mobile phones, as it allows you to sync with their office phone as well. http://www.plantronics.com/north_america/en_US/products/cat29880043/cat2 9880054/prod5460016

Snom 200 has be set up with extended key pad. The product literature also mention multiple sip registration. How many registration can it handle? It does not seem to appear in the user manual. David Kwok

Can someone explain what this error is? -- Got SIP response 500 "Server Internal Error - Invalid CSEQ number" back from 209.xxx.xxx.xxx How do I fix this? .o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office

has any one implemented asterisk with 723 and 729 codecs, what is the cheapest way. is there a limitation in the open 723 implementation ?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050225/d5daf369/attachment.htm

Hey all again, I have successfully compiled and am running Asterisk (stable release) on OS X (10.3). However, any make directive that uses the "-shared" option in gcc results in an error. Apple states that -shared is not supported under OS X. Is there a workaround or do I have just have to live life without those modules (zaptel, libpri, format_mp3, probably others)? Thanks, Zach

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

> -----Original Message----- > From: Iassen Hristov [mailto:ih.ng@databrokers.net] Dumb, hacky idea...but just so crazy it might work: Have Asterisk include a read receipt request when sending the voice mail message. Write a script, triggered from a sendmail alias or .forward file, that will parse the incoming receipts and handle the message deletion. Bonus points: When someone listens

Hi, When a SIP client registers on Asterisk server, why it expires after certain amount of time?

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

Hi We are in a project where we will use asterisk as a residential gateway for IP phone service. We are aiming to replace the primary phone line so the service must be up as long as possible so we are looking at ways to avoid shut downs. We are looking for a solution to allow us to add/remove PRI cards without shutting down the system Is there such a thing as an asterisk compatible

Hello there, I am playing around with Asterisk the first time and it really looks great. ;-) However, I have one problem: Any SIP device can connect to my PBX. How can I requre logon for SIP users and deny access in the case of wrong or missing credentials? Thanks Florian

Hi, I have several SIP registrations on my Asterisk box. Sometimes, I try to call in the inbound number from 1 and find it doesn't work. When I do sip show registry, it's showing Unregistered (and sometimes there are several which are showing Unregistered). If I type reload, it registers and works fine straight away. Is there something I can set to keep registrations connected all the