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Hello, I am installing Puppet Enterprise agent (puppet-enterprise-1.2.1- solaris-10-sparc) on a Solaris 10 64bit SPARC system. When I view the logs for the agent I see: puppet-agent[14680]: [ID 702911 daemon.error] Could not run Puppet configuration client: Could not retrieve local facts: private method `split'' called for nil:NilClass The client successfully initiates a cert exchange

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and

Hi, When I execute *puppet agent --test --debug --verbose* on the target host, I receive the following error message: err: Could not retrieve catalog from remote server: Error 400 on SERVER: Could not find class pe_accounts for targethost on node targethost warning: Not using cache on failed catalog err: Could not retrieve catalog; skipping run I enabled debugging for the Puppet Master, and

On Tue, Jan 27, 2015 at 5:53 AM, Charles Lepple <clepple at gmail.com> wrote: > On Jan 24, 2015, at 12:45 AM, Ryan Sizemore <ryan.sizemore at gmail.com> > wrote: > > Hello, > > I am trying to get nut running on a Windows 2012 R2 server (x64). I am > using the MSI release of 2.6.5-3. > > > I am not sure why they are not listed on the main download page, but there > are actually three W...

After browsing through bugs.digium.com I saw no mention of any work to get chan_sip or chan_sip2 to listen on tcp, as well as udp. Just curious is any-e-one working on such a patch at the moment?

Ex-girlfriend logic not working in latest CVS? Incoming sip calls don't work. Anyone else seen this problem? Extension logic looks good: exten => 6153248305/_931NXXXXXXX,1,Queue(queue1); exten => 6153248305/_615NXXXXXXX,1,Queue(queue2); ;exten => 6153248305,1,Queue(queue3); show dialplan looks good: -- Added extension '6153248305' priority 1 (CID match

...upsdrv_updateinfo... 34.110249 Got 2 HID objects... 34.141355 Full update... I'm not sure if the 'Read error' is indicative of anything, but it occurs exactly when upsd crashes with the 'Out of memory' error. Any help here would be greatly appreciated. -- Ryan Sizemore -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.alioth.debian.org/pipermail/nut-upsuser/attachments/20150123/94ecefcd/attachment.html>

On Jan 24, 2015, at 12:45 AM, Ryan Sizemore <ryan.sizemore at gmail.com> wrote: > Hello, > > I am trying to get nut running on a Windows 2012 R2 server (x64). I am using the MSI release of 2.6.5-3. I am not sure why they are not listed on the main download page, but there are actually three Windows MSI releases after that:...

...this in tests: [root@puppetmaster puppet]# cat modules/tomcat/tests/init.pp node default { include tomcat include create_resources include hiera } I don''t understand why my class doesn''t find the other classes. Can anyone shed some light? Thanks, Berry Sizemore -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to puppet-users@googlegroups.com. To unsubscribe from this group, send email to puppet-users+unsubscribe@googlegroups.com. For more options, visit this gr...

...;service httpd start') I get an error that the service command is not found. Why would that be? Where does this command come from? I can launch the services just fine using the respective /etc/init.d/ entries. -- Cheers, Trey ---- Dieters live life in the fasting lane. Linux fedora7.thesizemores.us 2.6.22.1-41.fc7 i686 GNU/Linux 16:51:36 up 1 day, 5:56, 2 users, load average: 0.08, 0.02, 0.01

Maybe its just me but I find this question a little confusing, the tone duration should have no impact on tone recognition and typically in my experience the duration of the tone is defined by how long the user holds down the button !? > -----Original Message----- > From: James Sizemore [mailto:james@deny.org] > Sent: 22 August 2003 17:33 > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] DTMF tones not long enough on out > going calls > > > DTMF tones are not long enough on out going calls, when I'm > using either > "info&q...

I would like to setup a hunt group, not a group ring, using sip phones. Anyone done this with sip devices? Comments suggestions? I have not had much luck with the outgoinglimit=1, incominglimit=1 stuff that I would need to get busy extinctions to work right, which is why I'm asking on the list.

I have a PRI that if you dial a number that is busy, the channel does not hang up, it then sends "h|1"to the phone company which will then plays back to the end sip user "You don't need to dial a one or zero" I am running stable CVS-v1-0-01/20/05-02:45:17. I have placed the important bit from the extension and sip configs below. Simplest possible example that will show

I am going to change my Digest realm to match my DNS SVR record. I dug through the code in chan_sip.c and on line 2748 I found it hard coded <frown> : snprintf(tmp, sizeof(tmp), "Digest realm=\"asterisk\", nonce=\"%s\"", r\anddata); I'm going to change this to : snprintf(tmp, sizeof(tmp), "Digest realm=\"isdn.net\",

Hello all, I noticed that Cisco claims that you can do station to station intercom with the 7940/7960 phones, and the Cisco Call Manager. Does anyone have an example SIP header that shows this in action? Or is it something else that triggers the intercom? I would like to add this in to *. Thanks, Mike

If you add "t" to you out-going trunk "Dial" lines: exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@vocal||t) exten => _NXXXXXX,2,Congestion so that you can still use park to park a call or transfer the phones, You have a problem of not being able to use "#" on external IVR systems. Is there any solution to this problem? Other then training hundreds of users not to

Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum.

I have been testing Gastman and Astman with SIP calls. As I have no Zap phones, so I have a few question on what is normal behavior? When a call comes in and I have created extensions for all phones (example: Channel = "SIP\3846") Should the little lines connect between the pre-made extension or should they pop up temporary icons with no connection to the hand made extensions? The

Can a call be kicked out of a queue if it reaches a specific timeout? I don't see an obvious way to do this in either queues.conf or extensions.conf any pointers or patches to do this? <smile>

Grandstreams phones can't call out with the latest CVS, anyone know what the last good CVS date was?