similar to: Fw: perl database get

Hi Darren/Damien, Sorry for responding so late. Still hope we can get this sorted out. Yes I am indeed using PAM for ssh authentication and disabling priv seperation is a no-go for us since it opens up a security loophole. From what I can see in ptree and auth logs, when the child passwd process returns with SIGCHLD, the parent sshd process terminates. Sshd logs are as follows as requested at

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

Hi, I am attempting to connect to the blacklist database using PHPAgi and it always seems to hang. The code snippet I am trying is: $r = $agi->get_variable("CALLERID(num)"); $cidnum = $r["data"]; if ($cidnum < 1000000000) # No valid callerid. { exit(0); } $r = $agi->database_get("blacklist", "$cidnum"); if ($r["result"] ==

and a little more on fax config support ----- Original Message ----- From: "Tilghman Lesher" <tilghman@mail.jeffandtilghman.com> To: <asterisk-users@lists.digium.com> Sent: Monday, August 11, 2003 4:48 PM Subject: Re: [Asterisk-Users] Fax Handled > On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote: > > On Mon, 11 Aug 2003 15:15:08 -0500 > > >

"In computer software standards and documentation, the term deprecation = is=20 applied to software features that are superseded and should be avoided.=20 Although deprecated features remain in the current version, their use = may=20 raise warning messages recommending alternate practices, and deprecation = may indicate that the feature will be removed in the future. Features = are=20

hi how can I find current codec from an AGI scipt? roy

Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

This is what you show CBeyond. You have some vouchers there from CBeyond that will allow me to get paid from them not you. Chuck Coleman President CCI Technologies/CC Call Center/CSI Technologies Director of Managed Services for Gurus2go Cell 510-439-6501 Confidential Email: This email and any files transmitted with it are confidential and intended solely for the use of the individual or

Sprint Nextel is sueing vonage, voiceglo and theglobe.com for infringing on VoIP patents. Sprint Nextel claims to have about 100 patents on VoIP technologies. Does anyone know which ones this article is talking about, and if so does asterisk have any of those features? The reason I am asking is that the article is vague, Vonage uses a fairly standard codec set, I dont know about the others.

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Mohammed Firdosh Nasim Sent: Tuesday, March 15, 2005 11:08 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] How to register two SIP phones ( e.g. WindowsMessenger) from different subnet to * On Sat, 2005-03-12 at 07:42, Luki wrote: >

Hi folks, I came across this issue on both stock CentOS(v6.4) and Ubuntu(14.04 LTS) and was wondering if any of you have seen it. As far as I can tell this seems like a day-1 bug to me. PROBLEM: If I expire a linux user's password (passwd -e <user>) and then log in via ssh, it will prompt you for a password change. On changing the password successfully, sshd will drop the connection

gtk12-config --version 1.2.10 glib12-config --version 1.2.10 AZ -----Original Message----- From: Tilghman Lesher [mailto:tilghman@mail.jeffandtilghman.com] Sent: Wednesday, May 21, 2003 2:27 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] gastman segmentation fault when pressing 'enter' in a command win dow On Wednesday 21 May 2003 01:22 pm, Alex Zarubin wrote:

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>

I found a message in the archives from Cameron Palmer on 23 Dec regarding his voiceglo SIP configuration. Unfortunately (for me), the archive has his email address removed. So, Cameron -- or anybody else using voiceglo with their * box -- please reply to me so that I can get your email address and ask you a question about your setup. Thanks, Greg

Olle ?) aiming to unify logging, eventing, monitoring (AMI, SNMP, ...) APIs. I think that thread occurred when it was decided to include a version number in Manager interface. I agree this is an interesting idea ... The use case that made me ask this is here : I've got a running system which is working ok up to a moment it stops to dial out on ISDN-BRI spans (incoming calls are ok). When

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hi. I am using #include to include a file in extensions.conf. I have an agi perl script which modifies the #included file and then forces an asterisk reload with 'system("asterisk -rx reload")'; After the reload I use set_context, set_extension and set_priority to tell asterisk where I want it to carry on from which is line 2 of the following. exten => *80,1,AGI(bla.pl,

This works fine. Message: 26 Date: Tue, 7 Jun 2005 17:22:23 +0800 From: "Ma Zhiyong" <mazhiyong@sohu.com> Subject: [Asterisk-Users] te405p and dell poweredge To: <asterisk-users@lists.digium.com> Message-ID: <004501c56b42$6a12fcd0$7301a8c0@hylaclient> Content-Type: text/plain; charset="gb2312" Hi, I want to install * and te405p on Dell Poweredge 1850. Can I

Hi. Can anyone help me with the following: [globals] OFFICEHOURS .................................... [internal] exten => *80,2,SetGlobalVar(OFFICEHOURS=100) exten => *80,2,SetGlobalVar(OFFICEHOURS=200) .................................... [incoming] exten => s,1,GotoIf($[${OFFICEHOURS} = 100}]?incoming-officehours:incoming-officehours-off 1. Am I using the right sytanx when

Is anyone signed up with Vonage and using an asterisks box?? Also what VOIP providers would anyone recommend? -- James Moran Potential Technologies http://www.potentialtech.com