similar to: Random Asterisk deaths

Anyone here running asterisk on openvz, if so what are your experiences? Right now we are trying to tune out the resources for the difference VEs, but not with a whole lot of luck. Just wondering if someone watching could shed some like on what has worked for them, and how many exts/simultaneous calls etc are happening. Thanks Miles -------------- next part -------------- An HTML attachment was

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

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'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 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 =

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 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

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:

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.

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

I pulled a working TE405P from one box and put it in another box. I compiled the zaptel modules, modprobe zaptel, modprobe wct4xxp, but no lights on the card come on. I do an lspci and the card shows up there. I ran ztcfg -vv and got the error message Unable to open master device '/dev/zap/ctl' so I followed the instructions in README.udev the error message went away, but now when I

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

I'm having various issues that may or may not be related to each other (I'm pretty sure they are). We've had this system for a year now (quad T1 card, right now we have 1 T1 coming in, 2 going out to channel banks) and we've had intermittent "ghost" calls--it appears that what is happening is a call is made, the number being called is disconnected/busy/answering

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

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

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

Greetings list, Does anyone have any experiences they'd like to share deploying these phones in medium-size asterisk setups, e.g. 40+ users? I have a project coming up to deploy 100 phones over 2 offices and the client rather likes these phones. Are there any obvious pitfalls/configuration difficulties/quality issues etc. using these phones? If so, what alternatives would people suggest with