similar to: Meeting

Hi all, We're having a meeting this coming Thursday, June 1, 7:00 at Pint on Punt (same as last time). Peter Fern has graciously volunteered to lead a discussion on ENUM and DUNDi: What they are, how they work, and how they work with Asterisk. Of course there will also be time to talk about phones and what's new in the Asterisk community. Date: Thursday June 1 Time: 7:00pm Place:

Once again, it's time for the Melbourne VOIP get together. Thursday the 7th is the December meeting of the Melbourne VOIP club. We will be meeting at the usual place which is Pint on Punt, 42 Punt Road Windsor. It's near the big messy intersection of Dandenong Road, King's Way, Punt Road, Fitzroy Street and St Kilda Road. Lots of trams go by there, and Windsor train station is just

I have a _very_ specific situation where I need queues to work in a very specific manner - I need the queue to only accept one call at a time, even though several phones are attached to it. My memory tells me that queues might have even worked this way in the distant past (pre 1.0)...but I am willing to be mistaken. Is this even remotely possible? PaulH

thankyou both very much for your swift responses and helpful insight... While my knowledge of administrating Asterisk is fairly decent, i must say my knowledgebase and ability in troubleshooting it is fairly lousy... all these wonderful suggestions you have had about turning this log on here, etc sounds great, but i don't know where to begin on that! how do you recommend i turn these on or

Is there a way to specify multiple email addresses in voicemail.conf for a specific user? I seem to remember that it was possible, but can't remember the character to separate the email addresses. (I tried '&', but that didn't work...) later, PaulH

I am trying to send 'codes' over an isdn2 link - such as *#24# - to activate call forwarding. But it doesn't work. I have tried sending it as a straight dial, and also as a DTMF string...but no luck... I spoke to a telco tech and he said I had to send a facility code....huh? Anyone with any ideas on this one? PaulH

Any idea how to read an external file, grab some stuff and push it back into an Asterisk variable? I can do it the other way with: system(echo "${UNIQUEID} =>" >> /home/ast/curr_calls) but I'm a bit stumped on how to go the other way around.... much thanks, Paul Hales

I have been discussing an asterisk solution with a company that has a custom written dialogic based solution. The issue is that their dialogic solution can read callerid from incoming calls, even if the callerid is blocked. I have read before that Asterisk can do this, and they want me to make sure that their new system will be able to do this. A quick poke around inside the zaptel source code

I need to carry a variable over into the 'h' priority - so I can go back and clean up DB entries in a mysql database (time of call and so on) I tried using UNIQUEID but it seems that 'h' generates a new one. Anyone have any ideas? What can I use to carry a variable over into 'h'?? later, PaulH

Does anyone know of any startups using Asterisk? What about established companies? Ones that are hiring would be nice.... :) Doug.

Be there and be square! PaulH -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of jurgen Sent: Thursday, 17 February 2005 8:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Melbourne Asterisk Users meet TONIGHT Hi all, Just a quick reminder: If you're in Melbourne

Hi, I explain what I want to do.. All the operators share their phones. The number of the operator isn't constant, so it's possible that two operators share all the phones. They need to move all around, so they pick up the first phone they find. If there are only few operator is very annoying for them to ear the other phones ringing while they are at the phone! So I'dd like to limit

Just wondering what web GUI people like for asterisk. I installed asterisk from source and I was looking at possibly installing web GUI for system management. So far freepbx.org looks promising anybody else have any suggestions. Thanks Roy Anciso Director of Technology Manistee Intermediate School District 1710 Merkey Road Manistee, MI 49660 Ph: 231-723-4264 Fx: 231-723-1690 roy

Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . Regards H.Motamedi --------------

Getting our own VOIP conference server going will be the 2nd part of the ProgrammingParty until it is accomplished. - The first part will be getting Ekiga ver 3 working on KUbuntu 8.04, & whatever other OSs people have. Come help out on the BTIP conference server, if you like. :) ===== Ekiga(Gnome meeting), Asterisk, Xen, Virtualbox, Debian 15 Years, Free and Open Future, Amarok, ZFS,

I'm guessing that I need more info entered into the 'message centre' section. What did you key in? Paul Hales IT Support Adairs -----Original Message----- From: Jeff Pyle [mailto:jpyle490@gmail.com] Sent: Friday, 17 September 2004 11:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP500 I have two IP 500's on my Asterisk

Hi Which is the best interface card to connect PSTN line with Asterisk. Can somebody please help. My intention is to route the incoming PSTN calls to internal IP Phones through Asterisk and Vice versa. The Asterisk is in LAN and is reachable from all the IP phones in the LAN. Thanks Manoj -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am trying to implement the callback feature of our old phone system. This feature may go by a different name in asterisk? It worked as follows. If phone A called phone B and it was BUSY, you press a button to enable a callback. User A is free to continue work or make other calls. What this meant is that when both phones became free, phone A would ring, on answer it would call phone B

Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

Hi there, I am looking to buy an Asterisk Appliance or Box for my organization and was hoping to ask for recommendations. My ideal box is a small device in size like Digium's AA50 Asterisk Appliance ( http://www.digium.com/en/products/appliance/ ) but will still have these technical features : -> Run Linux obviously -> Run a fullly configurable distro of Asterisk (not embedded) -> Can