Re: [asterisk-users] Asterisk on Xen or Dedicated
On 01/05/2008 00:27, "George Pajari" <[EMAIL PROTECTED]> wrote: > >> On Wed, 2008-04-30 at 13:11 +0100, Dee Lowndes wrote: >> >>> ...Question is do I still need to worry about timing and if so can this be >>> resolved in a Xen enviroment?... >>> > > We're an ITSP and use OpenVZ to offer customers Virtual Private Asterisk > Servers (see www.vpas.ca) -- the same idea as Virtual Private Servers in > the Linux world but with Asterisk added. > > Because of our network architecture, we chose to put the Digium cards in > dedicated (i.e. not virtualised) servers acting as gateways to several > OpenVZ servers so that the base environment (called VE0 in OpenVZ > nomenclature) does nothing but load the ztdummy module. All the client > VEs communicate with one or more SBCs or media gateways (i.e. servers > with Digium Quad-PRI cards) using SIP or IAX. > > Each virtual environment has access to a pseudo timer so they can run > meetme conferences etc. How does the pseudo timer compare to having a Digium card when handling large number of calls in a meetme conference? Also have you tried OpenVZ with a digium card does it allow direct access to it? > > Works very very well. We've migrated existing Asterisk configurations > from dedicated servers to OpenVZ virtual servers for customers who > cannot tell the difference. And a lot cleaner and more secure than > trying to run multi-tenant configurations/dialplans within a single > asterisk instance (which we still do for some customers for historical > reasons). I quiet like the sound of that as it does get a bit messy all on one asterisk instance. > > Sorry but we've no experience running Asterisk on Xen -- we looked at > Xen way back when were deciding on which way to go and chose OpenVZ > because it was (at least for us) easier to get running, easier to > support ztdummy, and more efficient (i.e. thinner) than Xen. > >>> One other question is how does multi cpu's scale is it better to have a >>> highspeed dual core or a lower speed quad core? >>> > > We use both and given the modest load you're proposing, it won't matter > -- get the cheapest. Our benchmarks showed that we get more bang for the > buck with X3210 Quad Core Xeons than the dual cores and so that is what > we've standardised on for now but YMMV. Thanks for the pointers. Dee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Xen or Dedicated
On 30/04/2008 13:15, "Steve Totaro" <[EMAIL PROTECTED]> wrote: > On Wed, Apr 30, 2008 at 8:11 AM, Dee Lowndes <[EMAIL PROTECTED]> > wrote: >> Hi All, >> >> I am trying to decide weather to move my asterisk setup on to a Xen >> setup or not. I do use transcoding, meetme and music on hold although in a >> purely sip scenario real lines are handled via cisco kit. Currently its a >> dedicated box with X100P card for timing handling it however it's starting >> to get a bit long in the tooth and I want to replace it. I currently handle >> up to 45 simultaneous calls but this will be doubling in the next year. >> >> Question is do I still need to worry about timing and if so can this be >> resolved in a Xen enviroment? >> >> One other question is how does multi cpu's scale is it better to have a >> highspeed dual core or a lower speed quad core? >> >> Any pointers greatly appreciated. >> >> Dee > > Just shooting from the hip but if you have the ability to use a > dedicated box for Asterisk, then go that route. > > Xen has a very high cool factor and may work OK with Asterisk but why > take the chance unless it is just a dev environment? > > Thanks, > Steve Totaro I see your point however I guess I just like to get the most use and flexibility out of my hardware. So having the 8 Core 8GB server I have setup only for Asterisk seems a bit overkill. I may end up having to purchase a more sensible spec for the asterisk box hence the question about dual or quad. Dee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk on Xen or Dedicated
Hi All, I am trying to decide weather to move my asterisk setup on to a Xen setup or not. I do use transcoding, meetme and music on hold although in a purely sip scenario real lines are handled via cisco kit. Currently its a dedicated box with X100P card for timing handling it however it's starting to get a bit long in the tooth and I want to replace it. I currently handle up to 45 simultaneous calls but this will be doubling in the next year. Question is do I still need to worry about timing and if so can this be resolved in a Xen enviroment? One other question is how does multi cpu's scale is it better to have a highspeed dual core or a lower speed quad core? Any pointers greatly appreciated. Dee ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 190, led and shared lines with asterisk
> In your extensions.conf create a hint:
>
> exten => 215,hint,SIP/215
>
> On the snom phone(s) subscribe the button to:
> destination:
>
> Where 192,168.0.200 is the ip of your asterisk server.
>
> When extension 215 is called, the light on the subscribed button on
the
> snom phones is light up.
I have exactly that
exten => 2,hint,SIP/2
exten => 2,1,SetGroup(${EXTEN})
exten => 2,2,CheckGroup(1)
exten => 2,3,Dial(SIP/sip-2,20,tr)
exten => 2,4,VoiceMail(2)
exten => 2,103,Busy
and
But all this achieves is a constant orange light with no signal if I
dial 2 from another phone. Also if I push the orange lit button on the
Snom it just rings extension 2 ideally I would like to be able to dial
from that extension but on the snom like a shared line.
Dee
>
> Netherlands.
>
>
> On Dec 17, 2004, at 8:37 PM, Dee Lowndes wrote:
>
> > Hi All,
> >
> > I am trying to setup my snom 190 so that the LED's light up when
> > one of my shared lines are in use.
> >
> > e.g.
> >
> > Extension 2 should ring on the snom and the phone associated with
> > extension 2 and I should be able to see if the phone associated with
> > extension 2 is making a call on the Snom.
> >
> > I think this is achieved with hints but don't seem to be getting on
> > very
> > well with it, does anyone have an example or pointers.
> >
> > Dee
> >
> > ___
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>
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[Asterisk-Users] Snom 190, led and shared lines with asterisk
Hi All, I am trying to setup my snom 190 so that the LED's light up when one of my shared lines are in use. e.g. Extension 2 should ring on the snom and the phone associated with extension 2 and I should be able to see if the phone associated with extension 2 is making a call on the Snom. I think this is achieved with hints but don't seem to be getting on very well with it, does anyone have an example or pointers. Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and echo
Hi all, I am noticing echo/jitter problems when going sip -> asterisk iax (ALAW)-> asterisk pstn depending on the codec I use. Both ULAW/ALAW works fine on the budgetone and ata286 but g726 only works well on the budgetone. Ilbc just doesn't work well with broken speech and echo issues. SIP to sip works fine no matter what codec so I am thinking it's either IAX or transcoding causing the issue. Any idea's/ Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: ATA 286 how to make the phone ring
Hi all, Does anyone know if its possible to get an ATA 286 to make the actual phone to ring instead of just the ATA ringing? Cheers, Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Slackware 10.0/Asterisk 1.0 compile error
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Juergen K. Zick > Sent: 12 October 2004 18:44 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Slackware 10.0/Asterisk 1.0 compile error > > Hi Folks, > > I just try to get * 1.0.0 compiled on a SLACKWARE 10.0 box. * 0.9.1 did > compile and work without any problems. But now, I run into an compile > error > which I just can't get resolved. > > ZAPTEL compiles OK, LIBPRI complies OK, but then during compilation of > ASTERISK: If you compiled 0.9.1 on the same system make sure you remove all old source dir's, /var/lib/asterisk and that X is installed. I did this and it all installed perfectly well on my slack 10 system. Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Config
Hi All, I am testing out Asterisk with IAX between 2 machines on local IP addresses and I want one machine to act as an IAX gateway with the other connecting to it. Anyone know of or can supply me an example of how to do this? Cheers, Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New $89 VOIP phone
> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Chris Shaw > Sent: 17 August 2004 21:12 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] New $89 VOIP phone > > A solution to this very problem has already been discussed... in fact... > in > this very thread! > > There are several options (much better than plugging your PC into the > phone > I might add) one being to buy a small desktop switch... Another would be > to > install a switching wall plate like those from Cisco or 3com... Another > would be to think ahead and install another wall jack if your company > might > now or in the future use IP telephony... Cool problem solved, now can someone say something important about this phone like how does it compare with a budgetone, what is the mic like, any firmware issues etc etc as I think two Ethernet ports on a phone are the least important aspect. Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting up your own menu like voice mail
Hi all, Anyone know where/how I can setup my own menu to work like the voicemailmain menu. e.g. extension.conf exten => 888,1,mymenusystem exten => 888,2,Goto(s,6) then somewhere mymenusystem plays message and give options to goto exten 1, 2, 3 etc Thanks in advance. Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK PSTN and x100p
Hi, > I use telewest my self and i have it set up to use Kewlstart, it does > disconnect the call, but its only after the teleewest line plays a > ringing noise, and then the telewest woman says "the other person has > cleared". > That is exactly what happens with mine by any chance did you get caller id working with it? > HTH > > Welby > > Iain Stevenson wrote: > >> >> Well. if the Telewest line signalling is the same as BT uses it >> "should" work. When the call ends the Telewest switch should signal >> this with a change in the line power which the X100P relies on to >> disconnect. the call. You'll probably need to measure the line voltage >> to sort this out. If I find the voltage drop out can I configure the x100p to do it based on the new voltage drop. If so where and how? >>If you have access to a BT line it's worth trying >> the X100P on that. >> >> Iain No BT line unfortunately. Cheers, Dee >> >> >> --On Sunday, March 21, 2004 15:32:54 + Dee Lowndes >> <[EMAIL PROTECTED]> wrote: >> >>> Hey All, >>> >>> I am using an x100p on a UK Telewest phone line and appear to be >>> having >>> problems with end user hang ups. >>> >>> If I call my * from and phone line and let * pick it up when I hang >>> up the >>> mobile or whatever I am calling from * continues with the call as if I >>> haven't hung up. >>> >>> Was wondering if anyone else has had this problem and knows a way >>> around >>> it. >>> >>> Thanks, >>> Dee >>> >>> >>> ___ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> To UNSUBSCRIBE or update options visit: >>>http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >> >> >> >> ___ >> Asterisk-Users mailing list >> [EMAIL PROTECTED] >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK PSTN and x100p
Hey All, I am using an x100p on a UK Telewest phone line and appear to be having problems with end user hang ups. If I call my * from and phone line and let * pick it up when I hang up the mobile or whatever I am calling from * continues with the call as if I haven't hung up. Was wondering if anyone else has had this problem and knows a way around it. Thanks, Dee ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
