similar to: Asterisk scalability

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

Hello, I'm checking some Billing Software for Asterisk. In opensource I only know (the name, I haven't used) AstBill. What other software should I check with similar capabilities? Thank you! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

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 =

Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona

Hello, (I have asked it some time ago in Asterisk-es mailing list, so excuse me if anybody receive it twice.) I am trying to send SMS in Spain using landlines. It seems that app_sms.c only handles Protocol 1, but Spain and Italy are using Protocol 2. I have been searching in Internet without any results... anybody is sending SMS from Asterisk (or any method) using Protocol 2? (so, it seems,

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?

Hello, Fast question: from DomU I cannot ping Dom0, I only get 19:17:33.573370 arp who-has 192.168.10.1 tell 192.168.10.150 19:17:34.573421 arp who-has 192.168.10.1 tell 192.168.10.150 using tcpdump in Dom0. Why? --------------- Detailed question: I''m setting up a Xen virtual server. The environment is: -HVM -AMD64 -Debian Etch and using Xen from Debian repository (so Xen 3.0.3,

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

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

Hello, In a couple of Asterisks, after type "sip show channels" we have a lot of these: IP_PEER dst_number something 00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say "a lot" I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, --

Hello, I'm sniffing traffic between Asterisk and a Softswitch. I see that, in "Decline" SIP packages, there is a header called "Reason" and I would like to access to the content of this header from Asterisk. How I can access to Reason header content? I would like to access here using ASterisk 1.4 and 1.2, but if it's only with Asterisk 1.4 will not be a big problem.

Hello, Today we setted up a server that needs to use MeetMe but doesn't have any Zap hardware. So we need to use ztdummy (at least, this was our idea). Rarely: zttest is not working at all (100% bad, using zttest -v doesn't give anything, etc.). Of course, after load ztdummy, there isn't any background or anything. It is the same kernel (Debian Etch default kernel, 2.6.18) than

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

Hello all.. Using dtrace on solaris 10, i could investigate a performance issue with the sincronization of some files on a ZFS filesystem. I have started the follow rsync command (inside a gnome-terminal): /opt/sfw/bin/rsync -av -e ssh user@IP:/DirA/DirB . The current directory(.), was a ZFS pool with two SATA discs (mirror)... The performance was terrible. After some tests with raid0,

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

The client is travelling much of the time. Is there some way that he can use Port 80 so that the firewalls that he is behind won't block the connection? Any other hints or suggestions are very welcome!