Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
2011/7/9 Doug Lytle supp...@drdos.info Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.**php?page=news_itempx=OTY0OQhttp://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- __**__**_ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Great news ! Thanks for sharing ! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
Hi, just in case had someone tested a similiar setup with VM on KVM ? Luca On 09/07/2011 01:28, Doug Lytle wrote: Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange behavior with Asterisk TLS
Hallo there! 1. i have Asterisk 1.8.2.2 2. installed on Debian 3. MySQL used as backend 4. I have configured TLS openssl genrsa -out key.pem 1024 openssl req -new -key key.pem -out request.pem rtc in sip.conf tlsenable=yes tlsbindaddr=0.0.0.0 [i am begind NAT] tlscertfile=/etc/asterisk/certificates/hereismyfilename.pem tlsdontverifyserver=no tlscipher=DES-CBC3-SHA tlsclientmethod=tlsv1 And to one test client i added: transport=tls Then i tryed to connect with BRIA softphone with enabled transport TLS and there is nothing happening... the phone is not connecting anv there is no message in CLI. But when i try to connect thru UDP then CLI says: Device not configured to use this transport type Only with sip cinfigured to udp all is working. Where could be my dumb mistake? Thx for any suggestion! -- Best regards Matiss Jekabsons Procerto Ltd.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT: Google Plus
Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! :r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DB Driven IVR
Anyone has Experience ? On Fri, Jul 8, 2011 at 2:18 PM, G M gm.cu...@gmail.com wrote: I am using Vicidial and I am looking for someone who can help with DB Driven IVR. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
On 07/09/2011 02:59 AM, Krishna Sumanth Chava wrote: Hi Doug, May be you can try a PCI-E card that has a PRI port and asterisk on itself and eliminating the need to install asterisk. www.positrontelecom.com http://www.positrontelecom.com have these cards. Interesting, never seen those before. But 900 g.729 - g.711 sessions on a single V-600 card? Or 200 T.38 sessions? Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Scheduled Maintenance for Asterisk Project Services
On Sunday, July 10th, from 8:00AM to 9:00AM Central Daylight Time, the Asterisk project services will be unavailable for use, while a database server is reconfigured and some virtual machine resources are realigned. The sites affected by this maintenance will be: * issues.asterisk.org/jira * wiki.asterisk.org * code.asterisk.org * git.asterisk.org We apologize for any inconvenience this may cause. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Google Plus
On 9 July 2011 12:34, randulo rand...@randulo.com wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! Can you suggest a good way of finding/following appropriate VoIP/Asterisk people once on Google+? How do you then group them? Just in a Circle, or some other mechanism? Thanks, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
Good point, folks, and thank you. I don't know yet whether they'll using something that is really a DS1-over-IP but that's what it sounds like. Since this would be a new Asterisk install rather than a legacy PBX, I probably prefer to go with somebody who can do plain SIP/IAX trunking (and not eat 1.5+ MBPS on the net connection constantly). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme not prompting for PIN
I've just put into place an updated meetme server. I went from 1.4.20.1
to 1.4.42.
In testing, it would seem that dynamically created conferences will not
prompt for the PIN. I've read though the readme and even went as far as
reading the 1.2 to 1.4 upgrade document.
s far as I can see, there has been no changes in the way dynamically
created conference rooms are handled. This is working correctly on a
1.4.30 server.
Below is a snippet of my dialplan with console output:
Dialplan:
exten =
s-process,n,Meetme(${conference.room}|ciMDPs|${conference.password})
Console:
-- Executing [s-process@mysql-meetme:1] SetMusicOnHold(DAHDI/1-1,
conference) in new stack
-- Executing [s-process@mysql-meetme:2] MeetMe(DAHDI/1-1,
1000|ciMDPs|1242) in new stack
-- Created MeetMe conference 1023 for conference '1000'
-- Recording
-- DAHDI/1-1 Playing 'vm-rec-name' (language 'en')
-- DAHDI/1-1 Playing 'beep' (language 'en')
-- x=0, open writing:
/var/spool/asterisk/meetme/meetme-username-1000-1 format: sln, 0xb742cd40
-- User ended message by pressing #
-- DAHDI/1-1 Playing 'auth-thankyou' (language 'en')
It skips prompting for password and moves right into prompting for the
recording of your name.
Suggestions would be appreciated,
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] OT: Google Plus
On Sat, 9 Jul 2011, Steve Davies wrote: On 9 July 2011 12:34, randulo rand...@randulo.com wrote: Go ahead and lambast me for this post, it isn't specific to Asterisk, but: G+ has only been open at all for a week and I already am chatting with over 200 people who are into VoIP, Asterisk and all the rest of the stuff we here care about. If you don't care or are anti-social, fine. But you owe it to yourself to check it, because a lot of cool VoIP people are there and after all, Google themselves are doing some great stuff with VoIP, XMPP and video, and steadily moving towards open source. Come drink the Kool-Aid! Can you suggest a good way of finding/following appropriate VoIP/Asterisk people once on Google+? How do you then group them? Just in a Circle, or some other mechanism? I've just created a VoIPy circle - So I can then invite people I know into the circle by email address, and/or looking at someone else's circles and seing if they have something relevant in their summary tag and adding them into your own circle... (Or using their people search - e.g. for 'randulo' :) You can have people in more than one circle. Right now, it's a bit like a media-rich version of twitter with excellent filtering (the circles). I don't have camera/microphone/speakers on my PC, (got real desk SIP phones!) so haven't tried the audio/video chat yet, but the typing instant messaging type chat works just fine. I think Google are still slowly gating people into + though. I did have some invites, but seem to have used them all up now (google didn't tell me how many, the invite button just went away after a while!) I'd love to see SIP integration into it, so I can use my existing SIP toys with it. Gordon-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Auto Reply: asterisk-users Digest, Vol 84, Issue 15
I am out of the office on vacation through July 20th, 2011. I am checking email, and will get back to you as soon as I can. For urgent matters, contact: Angie Besse for Oracle Labs, MA, and Corporate Security Architecture issues. Tami Sisneros regarding Corporate Architecture Approvals. Craig -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme not prompting for PIN
On Sat, Jul 9, 2011 at 11:31 AM, Doug Lytle supp...@drdos.info wrote:
I've just put into place an updated meetme server. I went from 1.4.20.1 to
1.4.42.
In testing, it would seem that dynamically created conferences will not
prompt for the PIN. I've read though the readme and even went as far as
reading the 1.2 to 1.4 upgrade document.
s far as I can see, there has been no changes in the way dynamically
created conference rooms are handled. This is working correctly on a 1.4.30
server.
Below is a snippet of my dialplan with console output:
Dialplan:
exten = s-process,n,Meetme(${**conference.room}|ciMDPs|${**
conference.password})
Console:
-- Executing [s-process@mysql-meetme:1] SetMusicOnHold(DAHDI/1-1,
conference) in new stack
-- Executing [s-process@mysql-meetme:2] MeetMe(DAHDI/1-1,
1000|ciMDPs|1242) in new stack
-- Created MeetMe conference 1023 for conference '1000'
-- Recording
-- DAHDI/1-1 Playing 'vm-rec-name' (language 'en')
-- DAHDI/1-1 Playing 'beep' (language 'en')
-- x=0, open writing: /var/spool/asterisk/meetme/**meetme-username-1000-1
format: sln, 0xb742cd40
-- User ended message by pressing #
-- DAHDI/1-1 Playing 'auth-thankyou' (language 'en')
It skips prompting for password and moves right into prompting for the
recording of your name.
Suggestions would be appreciated,
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
I guess you could do it the old fashioned way until you open a ticket on
Bugtracker?
Sorry, not much help, but I believe that Bugtracker is the proper way to
report a bug.
Thanks,
Steve Totaro
--
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Re: [asterisk-users] New VirtualBox Beta Has PCI Pass-Through Support
On Fri, Jul 8, 2011 at 9:41 PM, Warren Selby wcse...@selbytech.com wrote: On Fri, Jul 8, 2011 at 6:28 PM, Doug Lytle supp...@drdos.info wrote: Can you say a Virtualized Asterisk with a PRI card! http://www.phoronix.com/scan.php?page=news_itempx=OTY0OQ Doug Not trying to start a war here, but I thought Xen VM server has been able to do this for a while now? -- Thanks, --Warren Selby, dCAP http://www.SelbyTech.com To my way of thinking, none of this is ready for production. I refused to even waste time labbing any of it up, but now it seems that after a couple of years of people testing and tinkering, we are on the verge of VMs rather metal boxen, in production. I am cutting edge but still use sound and proven hardware and software configurations. This stuff is bleeding edge, but I think the edge is finally getting dull enough to deserve some of my time. The more and disparate ways of trying to make something solid and ready for real world usage the better. Each failure is a stepping stone to success. http://thephantomwriters.com/free_content/db/j/the-steppingstone-to-success.shtml Just because one project can already do something doesn't mean that are are not other roadblocks yet to be seen. Thanks, Steve Totaro -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Meetme not prompting for PIN
Steve Totaro wrote: I guess you could do it the old fashioned way until you open a ticket I've been searching the Jira issue tracker and found a ticket: https://issues.asterisk.org/jira/browse/ASTERISK-16747 Not being familiar with the new Jira system, I can't seem to find a patch for the 1.4 series, if I'm reading it correctly, the fix only went into 1.8. Can someone review and confirm? Thanks! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] About recompile reinstall couse of SRTP
Hi there! My issue is that i have (bouht to me) a box with Asterisk 1.8.2.2 but its seems that with no SRTP support. So i added a libsrtp libraries and like i understand now i need to recompile/reinstall Asterisk... is that safe to all data and MySQL backand data? -- Best regards Matiss Jekabsons Procerto Ltd. Dzelzavas Str. 117. Riga, Latvia-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
