similar to: Reasons why I shouldn't use Asterisk?

Hullo again, all :) If you're using * to run telephony in a real business environment, can I trouble you to write a short paragraph about the setup, and how you've found the migration / daily use? I'm simply trying to add weight to the business case for new * installs, especially for those who have a very conservative management structure. Like I say, I'm not looking for a case

Mm, I have a big Quad-Opteron.. thing.. that I''m trying to get Xen onto. I''ve used the 3.0.2 binary-install mode, updated menu.lst as per the README, but I need an initrd which contains the HP cciss RAID driver, and no Xen initrd image was installed into /boot. Now I notice xen-3.0.2-2-install/install/lib/modules/2.6.16-xen/kernel/drivers /block/cciss.ko But I

Afternoon/Evening all It looks like Puppet is starting to get some traction within my organisation, with several other teams asking what Puppet can do for them :-) The main use I''m getting asked about is one touch deployment of our products by Dev teams... On the surface it shouldn''t be too hard, and I''ve already got a model in mind. However one of the

Hi, I am doing a very informal presentation for my office about R capabilities to deal with and analyze spatial data, display data and maps, and connections with GIS. I've used in my presentation info from the CRAN, the spatial Task view, and the more striking graphics examples from http://addictedtor.free.fr/graphiques/thumbs.php and NCEAS

Hi :) I've had all this working before, but I'm revisiting it, and in short, I currently have huge problems receiving incoming calls. I've been trying with both FWD and voiptalk.org. I'm running CVS HEAD of asterisk, zaptel and libpri as of yesterday afternoon. Would someone mind helping? :) My machine is 10.0.0.1 on my LAN, but the ADSL router has 10.0.0.1 set as the 'DMZ

I known. I'm using the 1.44 firmware version relesed on 26 may. I worked for italian IVR an HTTP pgaes. So i can only update asterisk with CVS and try atxfer. Thanks for all -----Messaggio originale----- Da: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] Per conto di Gavin Hamill Inviato: luned? 30 maggio 2005 18.40 A:

Good...it almost works fine! I just have 't' in command Dial, but i also have 'T'. Is it a problem ? This is my Dial() exten => 605,1,Dial(${GIORDANO NAT},60,Ttr) I have only a problem: A and B are speaking, B calls C and ask it if wants speak with A, C accept but if B hang up A is waiting and C get busy tone. To make it works B don't have to hangup but habe to press

Today is my first day with sieve, so be gentle :) I'm trying to set up a pretty webmail interface to our Dovecot 1.2.4 server using roundcube. The managesieve config + roundcube 'managesieve' plugin work fine, and I'm able to use roundcube's UI to generate .dovecot.sieve files. We use winbind + LDAP lookups to do some exotic mail rewriting... ultimately user.name at domain.com

Hi, Thanks for yuor answer. The boot time of the phone is very very fast, 10 sec to startup and 2 or 3 second to login to asterisk. I set the NTP server to 255.255.255.255 so it don't try to get time. I thinked carefully to your scenario and i am going to try it, but i don't known if it could like to my customer I will try also to use CVS, but i am skeptic to utilize asterisk to

Hi :) My employer is looking to move a call centre to a new office, and has been increasingly frustrated with their legacy PBX (call-logging licensing and hardware upgrade costs). So I've stepped forth as the Open Source Pedant and suggested Asterisk so we can do all our own CallerID / call logging / analyses, and make use of IP Phones / teleworking, etc. The problem begins in that I only

Hello, I recently started playing with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives. 1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this? 2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the

Hullo :) Can someone help me untangle a bit of a mess? I'm trying to set up a demo * server to show off how useful it can be to our business (as an IVR system and VoIP backup if our ISDN30s fail). I've not been able to get NT mode working with our InterTel Axxess PBX, so I've resorted to using normal TE mode and working on the basis the people dial one of the ISDN BRI extension

Hullo. Having found http://bugs.digium.com/bug_view_page.php?bug_id=0000875 I grabbed the original 0.0.1 and Dan's patch, and whilst it didn't apply, I was able to patch the zaprtc.c manually - the Makefile has changed a lot, and I wasn't able to understand the changes. (this is all on a machine that's never had any * on it before, and has a 2.6.5 kernel with a matching

Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103's phone says 102. I've been looking for a way to have 103's Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the

Hi, First of all thanks for all the answers .... all were very informative and helped me shape a better presentation. Although i received lots of emails saying that installation under Linux / Unix is easy .... still emails asking for help in installing R itself, or external packages and such crop more often than not on the list and for this reason i pressumed it is harder to install under Linux

I am looking for pros and cons on the Intel Atom cpu. Has anybody been using these in production? I am looking at an Atom D510 (dual core 1.6GHz, 1M cache) to run maybe 25 to 30 extensions, 4 or 5 calls at once(maybe as high as 7 simultaneous calls), g729 all the way through except voicemail will be wav format for email purposes(requirement). I will be tying 3 of these together to route

Hi, First off, a big thanks to Digium (Mark, John, and Martin) for helping sort out a BellSouth config issue on our PRI. T100P working like a champ! Now it's back to tweaking the configuration on our SIP phones (7960s). The message_uri parameter in the phone's configuration file is working great. Dials comedian mail directly. Is there a way to let voicemail2 know what the incoming

Nothing much to be said.. I backported ringinuse, autofill and the QueueLog application from 1.4.0 to 1.2.14. Any or all may be applied - order doesn't matter. They have received minimal testing but appear to function correctly. As always with these things, don't blame me if they connect your callers to a phonesex line, etc. http://bum.net/patches/ Cheers, Gavin.

OK we talked about this a while ago, here's a summary and some proposals: At the moment, virtio PCI uses IO BARs for all accesses. The reason for IO use is the cost of different VM exit types of transactions and their emulation on KVM on x86 (it would be trivial to use memory BARs on non x86 platforms if they don't have PIO). Example benchmark (cycles per transaction): (io access) outw