search for: servat

Hi All I've compiled Samba 3.0 alpha 17 using --with-ldapsam. In smb.conf I've got: passdb backend = ldapsam ldap admin dn = "cn=Manager,o=Sambatest,c=AU" ldap suffix = c=AU ldap ssl = off ... and the admin dn password is in the secrets.tdb file. Whenever I try to connect to samba, I get the following error message: [2002/05/28 09:34:07, 5]

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?

Hi All, This topic has come up before in the Asterisk mailing list many times, so I know that a lot of people have given up in waiting for a FXO card to be approved by the Australian telecommunications authority. My question is: all legalities aside - is anyone using a FXO card in Australia successfully? Thanks in advance. Regards, Gonzalo

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

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...

...ms that demo run fine. I've also tried to configure with --enable-shared, but I receive the same error (core dump). Can anyone tell how I could solve this problem? Best regards and many thanks, -- ________________________________________________________________________ Harald Servat Gelabert (harald at cepba.upc.es) o//o Centre Europeu de Paral.lelisme de Barcelona CEPBA o//o WWW...: http://www.cepba.upc.es Tel: +34-93-401 74 23 o//o e-mail: suport at cepba.upc.es Fax: +34-93-401 25 77 o//o CEPBA c/Jordi Girona, 1-3, M?dul D6....

...t on R when quitting "q()". I'm using a newer version of R but on the same operating system and I obtain the same error. There was a way to solve this problem? Best regards and thanks, -- ________________________________________________________________________ Harald Servat Gelabert (harald@cepba.upc.es) o//o Centre Europeu de Paral.lelisme de Barcelona CEPBA o//o WWW...: http://www.cepba.upc.es Tel: +34-93-401 74 23 o//o e-mail: suport@cepba.upc.es Fax: +34-93-401 25 77 o//o CEPBA c/Jordi Girona, 1-3, Mòdul D6. E-080...

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 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

...uest@lists.digium.com You can reach the person managing the list at asterisk-users-admin@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: FXO Cards in Australia (Gonzalo Servat) 2. Re: IAX2 connectivity problem (qualify=yes) (Philipp von Klitzing) 3. RE: FXO Cards in Australia (Bryan Nolen) 4. RE: FXO Cards in Australia (Gonzalo Servat) 5. Re: Meetme : Zaptel ztdummy errors (Andrew Thompson) 6. SIP soft phone registration (Steve Murphy) 7. Re: DTMF (Sean...

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

...--without-blas As our machine and our user plans to use R only for a compute intesive calculations there's no need to use X. So this compilation is perfect for our purposes. Many thanks to all!, -- ________________________________________________________________________ Harald Servat Gelabert (harald@cepba.upc.es) o//o Centre Europeu de Paral.lelisme de Barcelona CEPBA o//o WWW...: http://www.cepba.upc.es Tel: +34-93-401 74 23 o//o e-mail: suport@cepba.upc.es Fax: +34-93-401 25 77 o//o CEPBA c/Jordi Girona, 1-3, M?dul D6. E-080...

...within that first week. But that's hardly a valid measurement, unless you're comparing against Windows Millenium systems. I call scam. --J -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Gonzalo Servat Sent: Friday, March 21, 2008 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ----www.cdsportal.net---- wholesale voipprovider --starting at 1.1 cent per min I think this type of abuse is well deserved due to the way he intended to advertise his &...

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

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, I would like to set two separate asterisks to talk to each other. Any suggestions? I'm a "baby" asterisk fan, only started to play two weeks ago, first managed to use kphone with asterisk and a X100P card that is up and running as well. Thanks, Eugen Find local movie times and trailers on Yahoo! Movies. http://au.movies.yahoo.com

Hi, is there a way to retrieve a VM message pressing some key during the greeting playback? Our scenario is a PBX with analog trunks and no DID. There's a general mailbox and no way to assign a number to voicemail. I've seen these question before in this list, but seen no answer to it... Thanks, Renato

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

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, I have 2 SIP clients configured and connected to Asterisk. When I place a call from SIP1 to SIP2, if both codecs are the same then everything works as expected. I then allowed one of the clients to use alaw instead of ulaw and there were audio problems (couldn't hear the other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw. Any ideas? I'm using