similar to: TE405P and E1

Hi All, I have configure my Asterisk as follow (using Asterisk@home): [zaptel.conf] span=1,1,0,ccs,hdb3,yellow bchan=1-15,17-31 dchan=16 loadzone = uk defaultzone=uk [zapata.conf] [channels] switchtype=euroisdn pridialplan=local prilocaldialplan=local internationalprefix=00 nationalprefix=0 usecallingpres=yes busydetect=no ; not need on pri callprogress=no ; was yes but wiki says experimatley

Hi, I am new with asterisk. I am setting a Wildcard TE405P. E1s in Italy come in on a pair of RG-59 coax cables with BNC connectors. So I need an adapter/balun http://www.allcomtlc.com/al_g703n3.htm . I have It but I am not sure that It works. I have configured my asterisk in this way: zaptel.conf span=1,1,0,ccs,hdb3,crc4 dchan=16 bchan=1-15,17-31 loadzone=it defaultzone=it zapata.conf

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, Digium's E400P has RJ45 conector and my E1 link has BNC concetor. Could someone tell me the cable pinouts to make this conection? thanks Eduardo

Hi, Question: Can a TCP server (running on the same host as the OpenSSH server) know the user id/name of a user forwarding an TCP port ? I.e. if someone on some client machine does ssh -L9999:localhost:9999 someuser at somehost nc localhost 9999 and a service accepts the connection on port localhost:9999 on somehost, can it somehow safely read out the user name "someuser"? Long

Im haveing a problem with my profiles share on my Samba 2.2.3 PDC server. I have a share like this: [profiles] path = /home/samba/profiles writeable = yes create mask = 0700 directory mask = 0700 browsable = no valid users = root,@smbusers The roaming profile works just fine with windows2k, and the users can't read the other profiles (they get a "access

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

Hi, does anyone have at working Perl example of how to use the auth protocol with Perl (I'm planing on making a dovecot auth plugin for qpsmtpd but it would a lot easier to reuse somebody elses code :)) tia, -- Allan Joergensen - http://nowhere.dk/ "All right, so it's impossible. How long will it take?" -------------- next part -------------- A non-text attachment was

No matter what I do, I'm not able to get window borders working. Both commands "emerald --replace" and "gtk-window-decorator --replace" runs fine, with no output, but no window borders are drawn. I've tried with and without: Option "AddARGBGLXVisuals" "True" ..defined in the Screen section of xorg.conf, but with no difference. To my

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

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 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

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

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

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

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