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file. I don''t know why it will work for other hosts but these steps do not work for this.. there are no extra outputs from using --debug On 4/9/07, Atom Powers <atom.powers at gmail.com> wrote: > > On 4/9/07, Mike Zupan <hijinks at gmail.com> wrote: > > I recently had a working puppet server serving around 4-5 clients. One > of > > the clients needed to

(yes, It''s horrible) IO.popen in terms of the non-blocking write functions implemented in ruby 1.9 or 1.8.5. However, that''s not necessarily such a great idea. Alternatively, you could use a signal-handler for SIGCHLD and still use IO.popen, but it seems difficult to ensure that you capture all of the output from the command up until the point that the process exits. You might

<br><br>Thanks in advance, best regards<br>Jose<br> ------=_Part_20741_28140847.1154122023605--

bind zones, virtual mailboxes or other mass-services. We currently don''t know how to handle so many vhosts through puppet. We saw that puppet can read node information in a ldap db, but puppet can only read NODE information, not vhosts parameters. As for today, is it (in your point of view) a clean way to do this with puppet ? We tried to found some way that may (or may not) work (some

(Correction: Subject was confused) Hi, I've noticed a possible minor issue with long encoded filenames for attachments where these filenames are split across multiple lines. My understanding of character encoding and MIME is not as good as it should be, so I may easily have got this all mixed up, in which case sorry for the noise... Although I understand the preferred method for handling

Hi, I've noticed a possible minor issue with long encoded filenames for attachments where these filenames are split across multiple lines. My understanding of character encoding and MIME is not as good as it should be, so I may easily have got this all mixed up, in which case sorry for the noise... Although I understand the preferred method for handling filenames split across multiple

sses, that way autoloading works ok and the classes are found, but that see= ms a bit awkward. <br></div><blockquote class=3D"gmail_quote" style=3D"border-left: 1px solid= rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br><br= >Note that it&#39;s a bit redundant to name your classes that way -- you<br= > can just as

Hi, I'm using Dovecot's (1.0.rc2) 'deliver' (no sieve) as LDA - calling it from a .qmail file. It seems that 'deliver' destroys "message/rfc822" attachments (forward as attachment). My .qmail file looks like this, and use qmail to deliver a copy directly to my Maildir, and 'deliver' to deliver a second copy of the same message to the same Maildir: %

Hi all, I''m having trouble with a particular recipe, but I don''t know which one! I''m not getting an error message, puppetd simply hangs (does not respond, needs a kill -9 to stop). These same recipes work correctly on other nodes (no problems there). Now I wanted to find the exact line that causes the problem, but I seem to be unable to make puppetd log the resource

Hi, I''m new to puppet and trying to get started using it just in local mode. My basic question is: how can i reference a static file within the module (eg in the module''s files/ subdirectory, as described in http://reductivelabs.com/trac/puppet/wiki/ModuleOrganisation). I suppose i could try using a full.absolute path to the file, but that seems to completely defeat the purpose

Hi, I''m a fairly new to Puppet but so far have been very pleased with the recipes and my own simple scripts. However, I''ve hit a wall in the form of OSSEC, (http://www.ossec.net/main/) Most of my servers are running ubuntu or debain and neither support OSSEC via apt-get. I''ve thought about setting up my own local repository to handle this and to also package my own

supported by Asterisk for Video. I also find that video_caps branch has a fix for this problem, please can someone share more information about this and where i can find it ? I do not want those fmtp lines to be stripped. Suggestions to change the Asterisk config files, to achieve this are also welcome. Thank you. Best regards, Simith ------=_Part_17870_6007467.1218041254938 Content-Type:

back to another ? Anyway, chan_dahdi.conf : [channels] language=fr context=isdntrunk switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 usecallerid=yes callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0

1. Is it normal to see : # lsmod Module Size Used by dahdi_dummy 3236 0 Shouldn't it be used by asterisk or is this 0 value meaning something specific ? 2. How can you check dahdi is running ? Here, "ps aux | grep dahdi " replies "grep dahdi". Cheers ------=_Part_2692_19661943.1228286635399 Content-Type: text/html; charset=ISO-8859-1

unusable. Is anyone using it successfully ? This kind of poll would be very useful to estimate is a code rewrite has a chance to disturb a running system. I we get no "successful report", it would help developers to consider a code rewrite as patch instead of a new feature. So please, do not hesitate to report successful or unsuccessful or never tried use of IMAP storage, it will

So I was thinking there's no need for a new codec. Am I right? Cheers, K ------=_Part_115139_19855101.1183563060148 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline <div>I&#39;m trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP).</div> <div>Problem is my video system only

*Update Jul 2007:* For a T.38 gateway you can use Asterisk 1.4's T.38pass-through support in combination with the new OPAL (Open Phone Abstraction Library) - using t38modem (currently CVS) which now supports SIP (and not just H.323) to terminate T.38 calls. You can also use OPAL and chan_woomera to do essentially the same. Where can you find this t38modem stuff ? Google replies things that

pages to mean "dial a new call to the mentioned phone number". Is there any pointer explaining available options ? I'm after something meaning "transfer ongoing call to the mentioned phone extension" instead of "dial a new call". Google replied me this : http://msdn2.microsoft.com/en-us/library/ms709071.aspx Anything else relevant ? Regards

Then, providing Linux compliant drivers should be Supermicro's problem. Anyway, how can you check in advance these compliance issues ? Maybe, a note in www.voip-info.org would avoid such issues. Regards ------=_Part_16952_4150825.1191306919994 Content-Type: text/html; charset=ISO-8859-1 Content-Transfer-Encoding: 7bit Content-Disposition: inline Hello Matt,<br><br>Do you mean

a features list I'm trying to translate and explain to prospective customers. I can't relate this Alarm receiver feature to anything meaningful. Does it mean anything precise to someone ? Does it mean you can connect an alarm appliance to an Asterisk analog interface ? Does it relate to Asterisk being able to exchange data with Linux system which in turn have many types of interfaces ?