similar to: How to find out the IP of the calling party?

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, 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.

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

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 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, 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 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

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. 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, 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

It there a native asterisk dialplan function which will tell me the position of a specific character in a given string? eg if I wanted to find what position the '@' was at in ${SIPURI} Thanks in advance Ish -- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w:

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, 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'd like to log the client IP addr and port used for SIP and RTP *during* in a call. The IPs must be the real source IPs (internet accessible). How are these parameters available from dialplan? For instance, ${SIPURI} holds the internal "IP:port" if the client is behind NAT. I need the external IP:port Regards Ethy

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 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

Hi All, I have a particular requirement where a Puppet managed group needs to have several members that are either local and not managed by Puppet (e.g. mysql) or they reside in LDAP. Apart from running an exec call to "groupmems", is there another way to achieve this? Thanks in advance. Gonzalo -- You received this message because you are subscribed to the Google Groups

Hi All, I was wondering what the status of distinctive ring support in Asterisk is? I had a google search & read and Mark Spencer wrote some support for it. Is distinctive ring different in every country or is it pretty standard? And for my final question, does the Wildcard FXO card support distinctive ring? Essentially what I'm trying to do is route incoming calls with ring #1 to,