search for: abarn

...ows box. Here's what I've tried already. If I try and mount one of the direcotries on our apple boxes I get an error message saying SAMBA alias could not be opened becasue the original item could not be found. If I try and modify a users password on the unix server I get ./smbpasswd abarnes ldap_connect_system: Binding to ldap server as "uid=admin,ou=Administrators,ou=TopologyManagement,o=NetscapeRoot" New SMB password: Retype new SMB password: ldap_connect_system: Binding to ldap server as "uid=admin,ou=Administrators,ou=TopologyManagement,o=NetscapeRoot" ld...

Has anyone had any success using the MixMonitor() plus "command" as nothing I have tried works. I am using 1.2.1 I did google the archive but couldn't see any mention of anyone using this. What I am hoping to do is run a macro on hangup, current method I am using seems to miss some calls 5% of calls fail to mix / convert to mp3 etc. Was hoping that MixMonitor would fix this.

Hi all, One of the areas I am trying to research before I can confidently start deploying Asterisk is "Credit Card Machines". (PDQ / Streamline machines / any similar) I'm talking about the credit card swipe boxes at point of sale desks. I believe they dial out to the specific bank provider everytime a card is swiped. My question is: - Does anyone have any experience using

Hi all, After many complaints (including car manufacturers saying the american prompts are unexceptable, EEEK) I started on a quest for real "English" asterisk prompts. The only one I have found is here >> http://www.g7ltt.com/VoIP/vmfiles.html <http://www.g7ltt.com/VoIP/vmfiles.html> And no nothing else on the WIKI looked helpful (e.g. only American voice actors etc)

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

-----Original Message----- From: Peter Svensson [mailto:psvasterisk@psv.nu] Sent: 25 November 2004 10:54 >> They work just fine if your pstn provider is at all serious. If not, >> switch. They don't belong in the pstn business anyway. BT so you probably have a point. To be fair to BT I'm not saying that the ISDN does drop during power outage I was simply speculating on the

> -----Original Message----- > From: Ronald [mailto:asterisk107@gmail.com] > Sent: 26 May 2005 10:47 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Little Php question > > > Hi > I'm trying to make a call from a local webpagee through my > xlite softphone > (xlite1) > BTW when I'm trying to do it through

Hi all, I have looked through the wiki guides and also Siemens user guides but they haven't proven useful. Nor has the normally trusty googling. Also have upgraded to the latest Optipoint 400 Standard SIP firmware. Having read a few previous threads on the Optipoint it seems that there isn't much take up with Asterisk. Which seems a shame as my experience with testing it has been

> -----Original Message----- > From: Michiel van Baak [mailto:michiel@vanbaak.info] > Sent: 26 May 2005 20:22 > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] CRM integration (was RE: CallerID) > > > Anton, > > My script is not connecting to the manager interface. > The php script is run as agi script as first when a call > comes in. The

> > > >| -----Original Message----- > >| From: asterisk-users-bounces@lists.digium.com > >| [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > >| Michiel van Baak > >| Sent: Martes, 24 de Mayo de 2005 04:45 p.m. > >| To: asterisk-users@lists.digium.com > >| Subject: Re: [Asterisk-Users] CallerID > >| > >| On 23:44, Tue 24

Hi all, Hopefully you can help me. I want to turn off the audio "Call Waiting" beep that plays during a call. I have found the line in the indications.conf for Call Waiting but apart from setting the frequency to zero or the length to zero is there a proper way to disable this functionality. thanks very much alex This email and any attached files are confidential and

> -----Original Message----- > From: David Liu [mailto:david@deltapath.com] > Sent: 31 January 2005 14:34 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Announcement to caller when > called party haspicked up - without initial Answer()? > > > This is super easy to do. All you need to do is to put that > announcement

Sorry I dont have any answers, however I do have a question. I was told that ISDN-30 lines do not work during power failure. Can anyone with some better knowledge confirm or deny this? Is this because the ISDN-30 box on the wall requires power (and Telco providers just dont hook them into UPS as standard)? Or do they mean if your local circuit has lost power so will the local digital exchange

Hi all, I have had quite a few customer complaints about attended transfer cutting off callers. The problem is when reception is busy she doesn't always wait for someone to answer the call, however hanging up a ringing transfer on attended also hangs up the caller. I have checked the scripts I don't *think* this is a dial plan error but if anyone has this working correctly on Asterisk

Hi, I am having difficulty getting my record phone call dial-plan script working. I have tried the example record call scripts but they start recording before they are actually connected to an end point, e.g. you get ringing or announcements being recorded. It seems to me that these are bugs with the Dial() command: 1) Using M(x) in a dial command does not allow argument to be passed. Using

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Rich Adamson > Sent: 08 February 2006 08:41 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Fedora Core 3 or Fedora Core 4? yum update > ornot? > > However, if you expose the box to

Hi all, I got a couple of Snom 190's through this week and after some initial foolishness I have them both setup no problems. But here comes the except. When there is voicemail waiting the softbutton appears but the phone always dials its own extension. No matter what I put into the "mailbox" parameter on the line settings. (Phone registers correctly with * and all standard