similar to: openvz resources

Hello, I wonder how Asterisk scales when we increment the Core's or CPU's of one computer. I see that Asterisk is only one process (I guess that it uses threads). But because Asterisk is only one process, this process is always executed in the same CPU. So we can have a 8 Cores server, with one Core running Asterisk, another Core running operating system stuff/other small daemons and 6

I have asterisk up and running on a fedora system but having trouble accessing system via softphone (ekiga and xlite). Im a newbie to asterisk and was looking for some help walking through this. I imagine 10 - 15 mins would be all needed to make proper config changes needed. Once I get this setup I'd be interested in discussing customizations and scripts so any freelancers or companies welcome

Anyone have opinions on how well Asterisk scales to 500-1000 stations, in regards to reliability, system performance, as well as ease of management? Assume relatively low call volume; let's say two public network PRIs (47 DS0s). -- # Jesse Molina # The Translational Genomics Research Institute # http://www.tgen.org # Mail = jmolina at tgen.org # Desk = 1.602.343.8459 # Cell =

Every once in a while for no apparent reason, Asterisk has been dying on me, dropping all calls in progress. There's nothing in the log file or on the Asterisk console that indicates the reason. Some days it doesn't happen at all. Other days it happens two or three times. The problem began on Friday, but the last time anything was changed on that box was at least a week before that.

Anyone had any experience with an Asterisk server as a VMWare virtual machine?

Hi, I've been asked for a headset recommandation for Polycom SoundPoint IP phones. Since I believe they use a pretty standard headset jack (correct me if I am wrong) it's really a general recommandation on headset. Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL:

I've been trying to get the polycom 550 phones to show a idle display bitmap but have not been successful. Anybody have any experience with this? The manual gives instructions (http://www.polycom.com/common/documents/support/setup_maintenance/products/voice/soundpoint_ip_soundstation_ip_administrators_guide_v2_2.pdf) but they do not seam to work. So far i've done the following in my

I have a customer that has two buildings that are connected with a fiber link. We have a single Asterisk server to cover both buildings. Now the customer went and bought an overhead paging system for the remote building and they want to integrate it with Asterisk. Is there a device that can connect over IP or an ATA that has an audio output port? The buildings are about 500 meters apart so we

All calls through the system are being dropped when they are bridged (Asterisk, Linux, pure VoIP system). The calling party here's half of the word 'hello' for instance and the call is dropped. I've noticed that hangup() was being called for some time now when the call was bridged, but the call was still continuing. Any thoughts on where to start debugging? Jason

I have a group of users whos complaint about Asterisk is that the directory application is too hard too use. (yeah, yeah, I know. For the record, they're Calgarians) Now I'm in a pickle: I don't want to have to create a custom directory for these guys. Anyone have any tips for making the directory easier, maybe re-record the prompts so they are more verbose? We go by first name.

May be slightly off topic, but I was wondering what everyone thinks of this latest ruling against Vonage? Does anyone really know what Verizon hold patents for, and could those patents possible affect anything in Asterisk? Who knows who Verizon will go after next. Brent -------------- next part -------------- An HTML attachment was scrubbed... URL:

I could have sworn I saw a post about this recently but I can't find it so apologies if this is a dupe, but is there anyway to control the order in the Polycom Buddies list? Bill -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070206/b2d3c1fc/attachment.htm

Tried to submit this email this morning and didn't see it in the list. I apologize if it is a dupe. I've inherited a customized Asterisk installation. After the past time change all clocks in my office are behind by one hour. After some digging it appears we have: A /tftproot/sip.conf that is being pushed out to our phones. I found the following line that seems to be what controls

Hi there, I searched in mailinglist and in web, but no answer to my problem... Only this post with no answers: http://lists.digium.com/pipermail/asterisk-users/2004-March/038994.html I'm using CVS Asterisk (05/17/04) with chan_capi 0.3.1. (multiple controller support). In my Asterisk-box there are 2 Fritzcards (module for second card compiled with changes on sourcecode found in the web).

hi all, how to establish a call between two asterisk servers for the sip users registered for the servers. ----- Original Message ----- From: <asterisk-users-request at lists.digium.com> To: <asterisk-users at lists.digium.com> Sent: Sunday, February 10, 2008 11:30 PM Subject: asterisk-users Digest, Vol 43, Issue 30 > Send asterisk-users mailing list submissions to >

I'm running into an issue w/ Buddy status on Polycom IP650 phones using buddy status (with SIP Hints) on Asterisk 1.4. Sometimes the status on the phones will "stick" in the busy status. I have noticed that I can call that extension & the status will reset (sometimes). Anyone else encountered this or anything similar. -Chris

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

Hello list, I was wondering if any of you guys have had any luck with polycom in remote offices, I'm facing a weird issue, polycom phones work fine in the main office, in remote office it says, Registration from '<sip:202@10.0.1.190>' failed for '70.59.21.112' - Wrong password the odd thing is Linksys phone works without any issue!! polycom wont register but its able to

Hi all, We are looking for information about Dynamic Realtime Asterisk, We have install some Snom phone 360 (SIP) for our customer , but we have a problem when we want to register 2 accounts on the same phone and on the same Asterisk PBX. The problem when we register two phone line in realtime it doesn't work, we can't make a call the registration failed when we place a call. Can

As you may have heard, Digium announced this morning that it's acquired Switchvox, a well known provider of Asterisk-based phone systems. Since several people have already asked me about the deal, I figured I'd let you all know my feelings on the matter. First of all, let me say that I personally think this is a great thing for all the parties involved. Obviously this gives Digium a