similar to: New User

Hi All; Anyone can advise for a good IP Phone that has the ability to support SIP firmware and IAX firmware? Ofcourse, SIP there is a lot, but we need also the ability to use IAX (as it is good for NAT). Any advise. Regards Bilal ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo!

Hi, I had a matrix with NULL values, which I wanted to replace with NA. Is there an efficient way to do this? Small sample input matrix: A B C D E 1 5222.18 6355.10 4392.68 2607.41 4524.09 2 NULL 257.33 NULL 161.51 119.44 3 NULL 274.80 305.28 443.27 NULL 4 1759.76 1556.45 1224.06 1558.73 1837.09 Tim

Sure thats all fine and dandy but i think its more of a secure option for the icecast2 server to handle this internaly than a 3rd party script. <p><p>Dave St John (CEO) Mediacast1.com ----- Original Message ----- From: "Scott Manley" <djsnm@djsnm.com> To: <icecast-dev@xiph.org> Sent: Monday, November 03, 2003 7:14 PM Subject: Re: [icecast-dev] Icecast User Login

The ticket clinic, What Is U.S. Copyright Law? The US Copyright Law grants rights to individuals for the works they create. The US Copyright Act of 1790 has changed over the years. The current basis of US copyright law is based on the Copyright Act of 1976. US copyright law is relatively automatic. Once someone has an idea and produces it in tangible form, the creator is the copyright holder and

Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: "Understanding the SIP, second edition" from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as

Hello I have few questions : - what's the difference between a subscribe request et a register request ? - in asterisk 1.6 allowguest=yes or no param does it work ? if yes, please someone could explain how doest it work because I think i'm a little bit confuse. - if I configure a sip terminal in sip.conf like this [john] type=friend username=JOHN secret=mypassword host=dynamic

I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald

Scott Manley wrote: > Actually if you;re interested in integration with some sort of > subscription services then you want as much of the subscription logic to > be external as possible - trust me - I designed a subscription based > system which handled 7 million users (and managed to bankrupt at least > one company ;-) > > > Basically when you get into subscription

I have a instance of Asterisk 1.2.14 that is being run from safe_asterisk. Asterisk is seg faulting and NOT generating a core dump. Why would that be? How can I make it dump core? Is there a setting in the safe_asterisk script that I am missing? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working with our SIP provider to nail down some call quality issues we're having, and they've asked me to provide SIP debug log files from our asterisk server. Is there a way to make asterisk 1.4 output only SIP debugging to a specific log file? Or it is best just to use tcpdump? Thank you! -- Jason Martin Metrix Matrix, Inc. 785 Elmgrove Road, Building 1, Rochester, NY

I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services > Voicemail on the phone. I also only get MWI events for whichever mailbox is listed

Hi List, While I know that "upping" ulimit will fix the issue I am trying to understand what will cause it. I have a few set ups that are almost exactly the same yet some machines used to give this error often and others don't. I also noticed the error a lot more on my boxes running 1.4.X. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Niklas - Thanks! Yeah, your Github issue was very useful for me to understand what is probably causing our issue (the syscall chain done on every UDP packet). Very interesting that you're able to see around 90% of a Gig line on bare metal. Were you ever able to make any further progress on adjusting Tinc based on the investigation in https://github.com/gsliepen/tinc/issues/110 ? Martin -

Hi, I had access to an hpc cluster, and wanted to parallelize some of my R code. I looked at the snow,nws, rscalapack documentation but was unable to make out how I should submit my job to the hpc, and how I should code a simple program. For example, if I had 10 matrices, and 10 processor how should I write the R (and the hpc submit code) so that I run the calculations (e.g. rowsums) for each

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

As you may have heard, Digium announced this morning that it's acquired Switchvox, a well known provider of Asterisk-based phone systems. Since several people have already asked me about the deal, I figured I'd let you all know my feelings on the matter. First of all, let me say that I personally think this is a great thing for all the parties involved. Obviously this gives Digium a

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

Hi, Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing high CPU utilization from the chan_sip module. I've notice the more sip peers I have loaded, the higher the CPU load goes when there are no active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4 Kernel 2.6.9-42.0.2.EL. I currently have 1558 sip peers loaded in Asterisk and the current CPU load is