similar to: Debian etch and web voice mail - how to configure it?

Hi, I wouldlike use webvmail on my asterisk, I use debain Sarge with asterisk 1.0.7 package. I have installed package "asterisk-web-vmail" but when i go to http://MyAsteriskBOx, i have a page of presentation of Apache. Could you help me please? Thanks

Appreciate any help - I am working through Agile web development (4.0) and am stuck on the deployment using phusion passenger. I get this: "Forbidden You don''t have permission to access / on this server." when I put depot.yourhost.com in the browser. I have been through various web searches and for file permissions and suggestions to change the <Files> container.

Hi, are there any common causes for a windows machines failure to find a samba domain controller? im trying to join a windows 2008 server to a samba[3.4.0] PDC and debug/netsetup says "failed to find a DC in the specified domain". cheers

On Sun, May 31, 2015 at 7:12 AM, The Doctor <doctor at doctor.nl2k.ab.ca> wrote: > So far BSD/OS and opensh 6.9 pre works with ZOC and Tera Term. > > Putty and WINSCP are broken. > Could you please elaborate on "broken"? Which version of PuTTY? (I'm not familiar with WinSCP versions but I believe the code is based on PuTTY, so I think if we figure out PuTTY then

>I'm thinking of how to backup an entire server ("server A") to a >remote area ("server B") using rsync. > >My initial thought was to have a shell account on server B, then have a >cronjob running as root on server A that uses rsync to send all the files >over to the shell account on server B. > >Then I thought that this is not reliable against

The next phase of my testing procedure involves the simple act of delivering mail to my test box. When I send a message to either a valid or relayed user at my remote machine's address, it never gets there. I know this virtually for sure because I'm tailing /var/log/maillog and nothing new has been added for the past hour. I'm thinking eventually my ISP will send the message back to me

I'm running into some errors I can't figure out. Situation: I have a Sparc IPX (2.6 Solaris), a Mac OS X box and an Ultra Sparc 1 running Solaris 9. I am trying to rsync a web site between the IPX and Ultra 1. All boxes are running 2.5.5 rsync built from source. Problem: Can't get a a sync between the IPX -> Ultra. IPX = kashmir Ultra = vortac Mac = archer Assumptions: a).

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi, I' ve just connected a carte X100M to my asterisk server running zaptel-1.2.5, libpri-1.2.2 and asterisk-1.2.6 on SUSE 10.0. When I make modprobe wcfxo and modprobe zaptel I haven't any error, I have also chan_zap.so module existing in /usr/lib/asterisk/modules. But, when i run ztcfg, it shows me this: Zaptel Configuration ====================== Channel map: 0 channels configured.

Has anyone had any experience with the Asterisk on a SuSe 10 platform? I'm currently using FC3 but because we use SuSe within other parts of the business I'm being pushed to changed the OS. Regards Lee ########################################### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/

Hi All... Can Asterick detect distinctive ringing on a POTS line and answer with different configurations? Thanks...

R-listers, Given a polygon and a circle defined by its center coordinates and a radius, I would like to calculate the area of overlap. I know that I can create a polygon from the circle and then use available packages to get the area of the intersection. However, because the polygon is of a fixed size and I will be doing this for circles of varying sizes, I'm concerned about

Yeah I saw that, which is what brought me to the mailing list. Ta From: Joe Julian <joe at julianfamily.org> To: Jonathan Archer <jf_archer at yahoo.com> Sent: Tuesday, 21 November 2017, 14:04 Subject: Re: [Gluster-users] Ganesha or Storhaug Not according to storehaug's GitHub page: > Currently this is a WIP content dump. If you want to get this up and running,

Thanks Kaleb, I presumed as much. I did previously try with 3.10 but ended up running into issues with pacemaker. I guess same question applied, can you point to any guides to a full setup of Ganesha on 3.10? Thanks From: Kaleb S. KEITHLEY <kkeithle at redhat.com> To: Jonathan Archer <jf_archer at yahoo.com>; "gluster-users at gluster.org" <gluster-users at

Hi, I would like to see the DNID in my MySQL CDR logs. I have read one big thread in the Asterisk Developer List, but I could not figure out how to implement it ? Is there a simple step-by-step. Thx in advance. Vai

All-- I'm attempting to implement something I've wanted for a while...a stdio link to a TCP port forward, at least for SSH2, but preferably for either protocol. There's certainly no technical reason this can't be done, but the vagaries of terminal / file descriptor handling are posing something of a challenge. Does anyone have any suggestions for "correct"

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can

Stopping and restarting Asterisk will lose the hints, then you will have to wait until the phone registers again. With 1.2.7.1 a reload shouldn't lose anything. Change the register time on the phones to something less that 60 minutes if it's a big problem. Instead of factory defaulting the phones you might find a simple restart will re-register the phone and BLF will work on the phone.