similar to: missing ring-tone

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.

Skipped content of type multipart/alternative

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

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

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

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,

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

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

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

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, 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

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:

Is there a way of temporarily suspending participants in a conference? Say I have 5 users, A,B,C,D,E and I wish A, B and C to have a discussion in the confbridge session that D and E can't hear, is there a way to suspend D and E for a while (whilst they are played music or whatever) and later join them back in? Failing that, I was considering kicking them and using an AGI script to rejoin

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:

Hi, how may I define in a callfile the CallerID presentation to be used for the requested call, eg. set it to prohibited? TIA, Bruno -------------- next part -------------- A non-text attachment was scrubbed... Name: Bruno.Voigt.vcf Type: text/x-vcard Size: 270 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050913/fcb5c595/Bruno.Voigt.vcf

Is there a way to flag a bad or noisy PRI channel in Asterisk so it will be skipped over? Shawn

Is it possible to build a dialplan in which shifting from daytime to nightime is not hour based but phone driven ???

Can someone tell me where I can get just app.c from. Mine somehow got corrupted, and no updates or anything else will fix it. I just need the one file from the latest cvs. 8-1-04. Please help