similar to: Large-scale Asterisk deployments: VON panel

Hi All. If anyone wants to help out with the asterisk track at the VON Developers conference, please let me know and I'll make the proper introductions. The 2004 VON Developers Conference will be taking place July 27-29 at the Marriott Pinnacle in Vancouver, Canada. More information is available at http://www.pulver.com/developers/ Best Regards Ed Guy

Having just returned from four days at the VON show in Santa Clara, I thought I would submit a "highlights" message. I hope others who attended the show will take the opportunity to add, as there was far more to see than I can cover on my own. [VoIP IS BIG] First, I have to say that VoIP is BIG. It is the buzz technology of the day. The show was packed, and everybody there was there

Anyone able to get me a comp/highly discounted ticket to this? $150 just to visit the exhibition halls sounds crazy? Dean > -----Original Message----- > From: Jeff Pulver [mailto:jeff@pulver.com] > Sent: Tuesday, 23 August 2005 11:47 AM > To: mailinglist1 > Subject: Register Today for Fall 2005 VON: "The Destination for IP > Communications" > > Hi There, >

I can't get my Asterisk to register/place calls with FWD. Here's what I have in my SIP.CONF: register => 11111@fwd.pulver.com/11111 [fwd] type=friend secret=somesecret host=fwd.pulver.com username=11111 fromuser=11111 fromdomain=fwd.pulver.com I'm using CVS version of Asterisk, checked it out last week. I get authenticate error when registering with fwd, and all my calls to

I have a small HUGE problem with *. I have installed * but I have 2 problems. 1 - Making call to FWD. 2 - Receiving call from FWD More info of the problem at the end. Here is the sip.conf file. ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = sip ;default Default for incoming calls register =>

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For all thos Asterisk users not on the FWD list, it works for me!: -----Original Message----- From: Free World Dialup - The Future of Dialing [mailto:FWD@LISTSERV.PULVER.COM] On Behalf Of Leonidas Piagkos Sent: donderdag 12 juni 2003 0:58 To: FWD@LISTSERV.PULVER.COM Subject: Re: [FWD] FWD losing Registration Hi Don, All you have to do with your ATA is to set the following parameters as :

Hello all - Second-to-last notice to collect RSVP's for interested parties for the Asterisk Birds of a Feather dinner/beer meeting. There are five people signed up for this at the moment. Looks like it's going to be the 23rd of September, since that fits well with the VON conference schedule. Please email me directly for details. JT >Date: Thu, 21 Aug 2003 01:52:52 -0700

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Hi, I''m currently helping out with a large scale rails application deployment. The idea is to have an apache 2.2 with mod_proxy serving out to a farm of mongrels distributed on N number of servers. We''ll also have some Starfish nodes doing processor intensive tasks distributed on N number of servers as well. I was wondering if anyone had any experience of managing and

Hi Everyone ! I have been using Asterisk for 2 to 3 months now. I have made peer to peer calls using Asterisk both with H.323 and SIP. Now there is a requirement that I have to setup asterisk in my office supporting upto a 1000 extensions (pstn or IP endpoints). Now the question that I wanted to ask was that is it possible to deploy Asterisk with so many extensions ? Has anyone ever used

I am seeing some very peculiar things in the routines that REGISTER my * server with several accounts. I saw this on my console: . . . NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout): Registration timed out, trying again NOTICE[5126]: File chan_sip.c, Line 1878 (sip_reg_timeout):

Astricon 2004 is the first Asterisk user's and developer's conference, to be held in Atlanta, Georgia in September. See http://www.astricon.net ** We will soon open for early bird registrations! ** To get a very low price, I recommend that you participate as a speaker. In fact, speakers doesn't pay any fee at all to participate in Astricon. **** Send us your speaker's proposal

Hi, up until now I've always deployed VMs with their storage located directly on the host system but as the number of VMs grows and the hardware becomes more powerful and can handle more virtual machines I'm concerned about a failure of the host taking down too many VMs in one go. As a result I'm now looking at moving to an infrastructure that uses shared storage instead so I can

I have recently started experimenting with Asterisk. I am running the system the other side of the a NAT router and trying to connect to FWD. I have opened UDP ports and have configured sip.conf to handle NAT. The problem: I can call from the FWD phone and the extension on Asterisk rings and there is two way sound so no problem. Now if in the extension.conf file I have, exten =>

Is this replacing Astricon this year? If so it looks like a pretty poor showing in comparison to Astricon Dallas last year. Cheers, Dean ________________________________ From: Carl Ford [mailto:carlf at vonmag.com] Sent: Wednesday, 18 July 2007 9:09 AM To: Dean Collins Subject: Announcing Digium/Asterisk World's Conference Program

2003-06-28 Bandwidth Study - John Todd (jtodd @loligo.com) Purpose: ------------- To obtain a better chart of actual bandwidth usage per codec as seen "on-the-wire" when using IAX2 trunking between two Asterisk telephony servers. Discussion: ------------- Past threads on the asterisk-dev and asterisk-users lists have indicated that the optimal way to save bandwidth on

Same thing. It will make sense to try Register => <FWDnum>@fwd.pulver.com:<FWDpass>@fwdnat.pulver.com:5082 but in that case Asterisk sends REGISTER sip:fwdnat.pulver.com SIP/2.0 which is not right. It should be sip:fwd.pulver.com but sent thru fwdnat.pulver.com:5082 BR Borut -----Original Message----- Subject: Re: [Asterisk-Users] Using Asterisk with FWD through NAT From:

I can't get the y axis to extend the full range that I need, which is -8 to 8 Here's my code. I tried using ylim, but it's still truncating at the extremes in my data. plot(x, y, type ="o", xlab="Panelist", ylab="T value",lwd=1.5,lty=1) xlim=range(1:14),ylim=range(-8:8), las=1) Any suggestions? Thanks! -- View this message in context:

Hi, My configuration: Sipura 2000 Debian/Sarge Asterisk 1.0.1 built by msp@toshiba on a i686 running Linux I am calling 612@fwd.pulver.com which is Daytimephoneline of pulver.com and for the first second the connection seems to be ok and I hear: <thu .. rsda ..> and nothing more, which suppose to mean <Thursday, ...>. The echo line from 613@fwd.pulver.com does work the same. I never