similar to: Temporarily placing confbridge participants on hold - two way muting

Hi, I'm writing a utility that needs to smbmount various shares from servers in numerous domains (no problem, all working) and then list the contents of the directories (no problem again) and obtain the windows file owner in a textual form..... Any ideas how I can achieve the last part efficiently? I see that smbcacls can do it 1 file at a time, I really need a way of doing it

On a zaptel TE410p, when a call is bridged PRI - PRI how much involvement does the processor have? We're now seeing chunks of missing audio and I can't tell whether this is due to a kernel upgrade or to a zaptel/libpri/asterisk upgrade. I'm not seeing missed interrupts (from a cat of the proc/zaptel files), any other ideas on how I could go about tracking this down? I'm

Would be nice to do both (type 2 and 3 I believe in Oftel terms), but I'd accept just our DDI if that was all I could get. Steve -----Original Message----- From: Storer, Darren [mailto:starusers@comgate.tv] Sent: 01 July 2004 09:35 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Providing Telewest in the UK with per extension outbound callerID Hi Steve,

As I understand it, you'd enter the extension at which you wish to be called back at, your 9665 has nothing to do with it. Instead of dialling 28 you could dial 9665 and that would add that SIP phone as an agent to the cytelcs queue. Steve -----Original Message----- From: Harold Workman [mailto:hworkman@cytelcom.com] Sent: 22 June 2004 18:54 To: asterisk-users@lists.digium.com Subject:

I'm having some trouble here - need some help! I've just bought a TE405p (32-bit 5V), but cannot get it working in a dell 6450. The dell has (from tech specs) "three peer PCI buses: two 64-bit buses and one 32-bit bus Expansion slots seven hot-pluggable PCI slots (two 64-bit/66 MHz, four 64-bit/33 MHz, and one 32-bit/33 MHz)" I cannot get the card working in any of the slots.

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Hi, ? I'm trying to figure out what the issue is splicing Asterisk between our Telewest PRI and a GDK-186 with a PRI card. ? We're using the Digium TE405P ? Our telco provider is Telewest, and Telco directly into switch is fine. ? When I splice Asterisk in, I can make and receive calls from Asterisk extensions, I can make outbound calls from the GDK, but inbound calls do not seem to pass

Hi there I've got a rather irritating problem with my Asterisk server. Whenever someone tries to call me, the don't get the usual "ring-tone" when they wait for me to pickup the phone. I don't know if I've disabled this feature somewhere in my configuration files. Since I'm in Denmark, I've got an entry in the indications.conf file pointing to

Hello, more and more operators in Europe offer music instead of ring tunes. E.g. instead of the 400 Hz or whatever tunes, the caller will hear J-Lo, or Mozart.... Currently I will have to answer the line to do that. Is there a way to do this with asterisk? Regards, Marc -- CTO Marc Storck MS Networks SA mstorck@luxadmin.org Internet Service

Ok, starting to get cheesed off and feeling rather silly. cvs head as of 5 minutes ago. #root asterisk -vvvvvvvc works, no problem. #root safe_asterisk works no problem #root service asterisk start Starting asterisk: [ OK ] #root asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) /var/run/asterisk.ctl and

I'm going to modify the queue announcements to allow for rounded seconds (e.g. we want to know to the tens of seconds. E.g. Average wait 1 minute 20 seconds). I'm going to add the optional announce of seconds to the queue config and a rounding factor (e.g. 10 in our case). The following parameters will be added Queue-announce-seconds (default is off) Queue-seconds (default

Hello all, I am trying to upgrade the firmware on my cisco 7910 without using CCM. I was told that chan skinny is possibly capable of doing that and would like to make sure. I have P00405000600 firmware which I have put in version in skinny.conf. the phone basiclaly stops at verifying load. tcpdump shows nothing happening apart from small amount of traffic to port 2000 (skinny). Does anyone

Is there any reason why there should ever be more than 1 instance of mpg123 running on a * server? I just did an 'uptime' and noticed all 3 of my loads where over 3.00. 'top' showed 8 mpg123 processes all processing the same 3 songs (our background music). I tried to kill one of them but another one spawned in its place. Any ideas? Thanks, Matthew

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Sorry to bore you more with the clock issue, but have you check /proc/zaptel/<span> to make sure it's not missing interrupts? There's also an option to record the audio for the fax, you could listen to that vs a recorded file that will receive correctly on a fax machine and see whether there is an obvious difference? (Good luck, that'll be really scraping the barrel!!) Does it

We're rolling out Cisco 7940 phones, linked to *, which is running a TE405p EuroISDN. We have 2 ISDN lines, one we had for testing, and one for general (40+ users) use. During the testing phase, we had 10 phones linked to the second ISDN line, and there were no problems with echo at all. Lucky me. However, since we have started rolling out, we've had quite loud complaints that there is

Alright, folks. I just deployed * into full production at my office. We have around 50 7905's, 5 7940's, and a handful of soft clients. We run a call center with around 15 agents. I also have a queue set up for the receptionists so that they don't get bombarded with calls. Everything seems to be working with a very few minor glitches. I firmly believe that the few problems we are

Maybe I over-complicated my question with the mailfax part. If I leave the mailfax step out entirely, then there should be a .tif file, right? But there's not. No tif file gets created at all. Permissions on the fax folder are 777 at the moment. Thanks for the responses so far. /Rob -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, We're using our legacy PBX as a channel bank with asterisk sitting between the pbx and our telco provider spliced by a TE410P. If it were a straight analog FXS card then we'd use a hook flash to break into asterisk for transfers etc, does anybody know what the equivalent is for the PRI zaptel support? Regards Steve Steve Hanselman Brendata (UK) Ltd Tel:

That sound's like the right thing to do, you'd probably have contexts related to what phones could do "unrestricted","localdialling","extensionsonly" and then within those contexts include the relevant contexts. If you look at the sample configs you can see how this is done for the international and local and you can extend that concept for your extension