search for: gservat

Hi All, In my set-up, I''ve got a cron job that triggers a Puppet run every 20 minutes. I''ve found that on approximately 13 nodes (out of 166), puppetd just hangs. I have to go in, kill the process, remove /var/lib/puppet/state/puppetdlock, and run puppet again and then it''s fine. After a while, it just hangs again so I have to go in, kill the process, etc. Any ideas?

If you write a HowTo, would you please insert it into the wiki at http://www.voip-info.org/wiki/index.php?page=LDAP ? Thanks. On Sun, 2008-03-16 at 07:09 -0500, asterisk-users-request at lists.digium.com wrote: > Date: Sat, 15 Mar 2008 18:20:32 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] LDAP > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <dcc007e10803151320s3498a7ecg9837e5a94a36389 at mail.gmail.com> > Cont...

Hi All, I'm trying to achieve the following: - If <sip/iax user> logs in from home, they can dial internal extensions only (this is to avoid employees going wild on local/mobile calls from home) - If <sip/iax user> logs in from the office, they can call anyone they want. Since I have my users defined in an LDAP tree, I'd like to stick to one-account-per-user (each account is

Hi, all. Shorewall Version: 1.3.12 I''m unable to list files (using PASV *or* PORT mode) on any FTP site that listens on a port other than 21 (from a client machine behind Shorewall -- from the Shorewall box I can list files no problem on the same sites) I have "Netfilter FTP" support compiled IN the kernel. Any ideas? My rule set is pretty generic. LOC -> NET Policy to

Hi All, I've just compiled Asterisk 1.4.18 and I'm planning on using an LDAP tree where the users will each have their account, SIP username/password, extension number, context, etc. My first question is: can this be done with 1.4.x? If so, where can I get the res_config_ldap from?? I googled quite a bit and found a res_config_ldap that looks to be coded for 1.2. Is anyone running

Hi All, This question is probably more for the LDAP experienced users/developers as I'm sure it would work fine if I weren't using LDAP (but I am, and I'm almost there with the setup!!!). I've setup an extension with the following: AstExtension: 210 AstApplication: Macro AstApplicationData: call-ext,SIP/testuser&IAX2/testuser,210 When I dial this extension, I see the

Hi All, In the example42 modules, I noticed the following is done: # Calculate OS version (without using lsb facts) $ossplit=split($operatingsystemrelease, ''[.]'') $osver=$ossplit[0] I''ve been using $lsbmajdistrelease in all my configs. Is there any particular reason why I might want to use the above instead? (apart from the dependency on redhat-lsb, which is

starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us support at cdsportal.net -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I've recently got a "second" number installed on my PSTN line, trusting the Asterisk distinctive ring detection would work as expected. It appeared to work fine at the start, as the second number generated a different ring pattern to 0,0,0 (in the console) only to realise that almost every phone call to this "second" number generated a different ring pattern.

Hi, It''s easy enough to modify, say, the "exclude" variable in /etc/yum.conf with something like: augeas { yum_exclude_kernel: context => ''/files/etc/yum.conf/main'', changes => ''set exclude kernel*'' } However, I''d like a more flexible system where I can "build" the list of excludes from

Hi guys, I'm interconnecting an Asterisk box with a Lucent Definity PBX by means of FXO/FXS ports on a TDM2400 card. Everything works well, except for one little thing. Every now and then somebody (from an Asterisk extension) will call another extension on the Lucent Definity PBX and they hit their voicemail. They caller leaves their message (or not) and hangup, BUT the Lucent sometimes

Hi All, I''ve downloaded a Puppet module and I''m trying to add a parameter to it by editing lib/puppet/type/<resource>.rb. I simply added: newproperty(:pcfree) do desc "My description here" end In the corresponding file in lib/puppet/provider, I do something with :pcfree. Whenever I call the resource from Puppet with my new parameter, it keeps

How do folks feel about getting Puppet job listings on this list? I''ve rejected a few that we quite spammy, but when the subject matter really is a system admin with puppet experience, the decision becomes a bit different. I''m looking for general feelings. A simple +1 or -1 would be great. Mike -- You received this message because you are subscribed to the Google Groups

...; atis at iq-labs.net > Skype: atis.lezdins > Cell Phone: +371 28806004 > Cell Phone: +1 800 7300689 > Work phone: +1 800 7502835 > > > > ------------------------------ > > Message: 20 > Date: Fri, 21 Nov 2008 13:46:00 -0200 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] Large Asterisk installarions (~10, 000 > extensions), preferably at universities > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID:...

...t; An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20080321/414c3d1b/attachment-0001.htm > > ------------------------------ > > Message: 5 > Date: Fri, 21 Mar 2008 13:52:17 -0300 > From: "Gonzalo Servat" <gservat at gmail.com> > Subject: Re: [asterisk-users] ----www.cdsportal.net---- wholesale voip > provider --starting at 1.1 cent per min > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <dcc007e108...

...then forced to go through VoIP, with the initial #9 not being passed to Asterisk. Adapt and use. :-) Hope this helps, Roshan -- http://roshan.info Be different, act normal. ------------------------------ Message: 2 Date: Sun, 23 Apr 2006 05:34:15 +1000 From: "Gonzalo Servat" <gservat@gmail.com> Subject: Re: [Asterisk-Users] Sipura SP3000 question To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <dcc007e10604221234o415d5009od0ab1720bac4ff6c@mail.gmail.com> Content-Type: text/plain; charset=ISO-...

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML

While I''m in temporary retirement, I''ve decided spend a little time experimenting with new things and making some updates to the web site. The biggest result of this effort to date has been: http://shorewall.sf.net/Shorewall_Squid_Usage.html This outlines how to use Squid as a transparent proxy running on the firewall, in the DMZ or in the local network. In the latter two

Hi, Has anyone else experienced problems as of the last couple of months when outbound calling through Freshtel? I've started getting a "No authority found" error. I've tried contacting them, and they seem to have some serious communication issues with their IT team, infact I think they have serious issues in their IT team full stop. First they can't find my account in

Hi All, I have a number of SPAX00X units (spa1001, 2002, etc) and about 30 odd PAP2-NA units all hooked up to Asterisk. As you can imagine, setting them up took a while, and changing settings on them also takes a while. In order to prepare for future deployments, I'd like to use XML provisioning (or any kind of remote provisioning). I figured since Sipura/Cisco won't release the utility