similar to: Phones won't stop ringing

We have a tenant who has been having issues with a congested connection and in trouble shooting it we've noticed that there seems to be a lot of SIP traffic even when none of the phones are doing anything. We've determined that this traffic is mostly INFO packets generated by setting qualify=2000. I understand that 2000 ms is the default value for the qualification parameter but what

I just wanted to post this so that it was out there and Googleable. Hopefully it will save other people a bit of time. If you have a Cisco phone (I was testing with a 7970, though presumably it would affect 7960 and others as well) that is looping trying to fetch the CTL tlv file - it may be because you are using Debians 'tftpd' (should be netkit-tftpd...*cough*hey, Debian

Recently (and for ages, I'm sure) folks have suffered partition destruction and had to try and recover. In the recent thread, the victim eventually had to resort to Google and fond some package that I can not remember now. Well, I was perusing my YumInfo.lst.05, for general info, and I discovered this (potential) little gem. Thought I would pass it on and make it "more googleable"

I have deployed several 1.8.3.2 systems as upgrades of customers systems and now I am seeing random crashes. For some reason the builds lock up and stop taking sip connections. Existing calls stay on but when the user hangs up no new calls or reg attempts work. In most cases a "core restart now" cleans things up. Some times I have to kill the asterisk process. The stability of 1.8.2

Hi all, I was just made aware of this: <http://www.codeigniter.com> I wonder where the inspiration comes from? Lots of very similar concepts there ;) I''ve only watched the intro movie so far, but very interesting. jt

Hi guys, I have one question: When refactoring the working code I need or should refactoring the specs too? My old code is: http://pastie.org/private/gpskgtavm4yzutanq1ro3w My new refactored code: http://pastie.org/private/2emgi1hr5iga3m9jib4q and the specs that are still passing without refactoring: http://pastie.org/private/qn3uaoin0k2n8vjsusssg Regards. -- "Most of you are

Hi, I''m running into a problem with the user type with puppet 0.20.1... this worked before with 0.19.x at least, not sure about 0.20. But I''ve got adduser.pp: <snip> define adduser($uid, $gid, $groups="users", $membership="minimum") { group { $name: gid => $gid, allowdupe => false, ensure

skype has been discussed here. This Friday, one of the subjects of our weekly Voip Users Conference will be about trying to connect our asterisk pbx with Skype. I have no nexus with Skype, Paypal or Ebay. In fact, (Note for google:) I have had recent serious ID theft problems resulting in fraudulent sales that Ebay is not helping me resolve - I hate Ebay - Ebay does NOT seriously fight fraud,

These patches (should) give us support for compiling against Ruby 1.9. Roy _______________________________________________ wxruby-users mailing list wxruby-users@rubyforge.org http://rubyforge.org/mailman/listinfo/wxruby-users

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Hi folks. postgresql fresh install, roles, users, databases, all done. I can log in after I SSH to the box. I entered my user (myself) in pg_hba.conf: host all rich 192.168.1.4 255.255.255.255 md5 When I nmap the CentOS box, I can?t see that port open. I?ve turned off SELinux, so I?m not sure why I can?t see the open port 5432. What am I missing? It has to be something

Dear all, This is just a placeholder really - I've spent quite a lot of time trying to track down the cause of a problem I'm seeing here, and have eventually got an answer. Thought it might be useful to have the solution googleable. Obviously, if someone has better ideas, I'd welcome comments. Similarly, if anyone wants detailed logs, tcpdump or strace output, I'd be happy to

Currently, facter returns the kernel version as both the kernel and operatingsystemrelease facts. I would like to change that so that on Fedora, operatingsystemrelease is the release number (5,6,7 etc.) or ''Rawhide'' ... are there any objections to making that change ? David

Hello list, I'm having some issues with some phones that don't stop ringing after the call is answered somewhere else. Basically, a call comes, enters a queue and all the phones in the queue ring. The problem is that when the call is answered, some phones don't stop ringing. I don't know if it is a configuration file, but I don't think so. queues.conf, sip.conf and

What purpose do the WaitExten()s serve here? Are you really allowing the caller to connect to different extensions in the test-queue context? Have you tried without the WaitExten()s? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150507/d97ed1a4/attachment.html>

James, The WaitExten()s just provide a pause between the two Queue() calls to let the first group of phones finish ringing. In this example I am ringing the same group (queue_level_1) twice, however in a real-world scenario I would ring queue_level_1 and then ring queue_level_2 which each have a different list of phones. Thanks, Andrew ----- Original Message ----- > From: "James

Just a "heads-up" ... my home asterisk server is being flooded by someone from IP 184.73.17.150 which is an Amazon EC2 instance by the looks of it - they're trying to send SIP subscribes to one account - and they're flooding the requests in - it's averaging some 600Kbits/sec of incoming UDP data or about 200 a second )-: This is much worse than anything else I've

> > Sorry, of course cdr.conf not queues.conf. marcus > > Am 01.12.2010 19:16 schrieb "marcus rothe" <synco16 at googlemail.com>: > > > Hi Rodrigo, have you got enabled the appropriate line in queues. Conf? > Regards Marcus > > Thanks very much, I include the line "unansweredy=yes" in the cdr.conf and solve the problem. Thanks again! --

Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia

for the googleable r-help archives, I thought I would post what I wrote into my .Rprofile to automatically set some system information. the most relevant aspect is the determination of mc.cores. this is useful when users want to use the parallel package options(uname= system("uname", intern=TRUE)) options(os= if (getOption("uname")=="Darwin") "osx"