Displaying 20 results from an estimated 9000 matches similar to: "(no subject)"
2004 May 13
3
recommend a Linux based TFTP server
Hi, can anyone recommend a Linux based TFTP server to go on an asterisk box?
Thanks in advance
Robb
2004 Apr 30
1
T100P & Integrated (D&V) T1 -> Public IP Range
Hello,
I have a T100P card on a Red Hat 8 machine with Asterisk installed on it and I'm trying to get the data routing properly through it.
I purchased an Integrated Data/Voice T1 Line with channels 1-6 voice and 13-24 data. The line is plugged directly into the Asterisk machine's T100P card.
Previous to the asterisk set up we had an Adtran router that connected everything to what the
2007 Oct 02
2
Having problems posting to the list
Hi All
I'm having problems posting to this list, no bounces the mails just
dont show
any advice how to get the postings through is there filtering?
robb
2004 Jul 12
4
call Intrude
Hi
I have looked through the wiki and search the mailing list, but I cannot
find a way to intrude on a call, can asterisk do this feature?
if so how?
Thanks for your help
Robb
2008 Nov 30
3
DTMF Tones
Hi All
I cannot seem to find a way to stop atserisk inercepting DTMF tones and
regenerating them even on a zap to zap bridged call
is this possible?
Thanks
Robb
2004 May 12
2
i cannot find kinit
the name of my active directory domain is
niit.edu.pk
so what should i write in this parameter
default_relam = YOUR.KERBEROS.REALM
also while trying to join the domain i eecute this
command
kinit Administrator@your.keberos.REALM
My shell gives me the error cannot find kinit.
can any one tell me where in my file system can i find
kinit
Regards
=====
Sahibzada Junaid Noor
Ph #
2008 Oct 06
1
R key with Siemens Gigaset IP (was MWI with Siemens Gigaset S450IP)
2008/10/5 robert.boardman at gmail.com <robert.boardman at gmail.com>
> Kevin P. Fleming wrote:
> > Olivier wrote:
> >
> >
> >> 2. R Hook-flash key is now available to transfer calls.
> >> In s450IP web management server, its defaults settings are :
> >> Application-type: dtmf-relay
> >> Application-signal: 16
> >>
>
2007 Apr 17
1
Transfercapability DIGITAL
Hi
I have a requirement to bridge Digital ISDN call through an asterisk box
but no matter what I setup in the dial plan the second leg of the zap
bridge is always set to Transfer Capability of SPEECH, I wondered if any
one has come across this and managed to fix it?
Thanks in advance for your help
Robb
2007 May 11
2
megasr Sata Raid driver and the lastest kernel
Hi List
I'm trying to update to the lastest kernel but I have a dirver that is not
inculded in the distrubution, and I had to use the driver disk when installing
centos 4.4 in the first place, The driver megasr .ko works fine with the
installed kernel but I cannot find on for the updated kernel, any adive would
be appreciated.
without the updated driver there is a kernel panic on boot due to
2003 Sep 12
1
Dect Phone
Hi
I have a problem with a new DECT phone I have bought
The key pad works like a Mobile phone where you dial first then pick up
the line, but it seems to dail too fast or spuriously, ie 012826736464
show on thew Asterisk console as 0012282677, could any one offer advice
how to fix?
Also when doing a ZAP bridge to this phone from an outside line the call
is very echoy, but not an internal
2008 Jan 13
2
problems with zaptel and Udev
Hi
I have had a Centos 5 installed with asterisk and zaptel for a couple of
weeks, I had to reboot eh machine today, and when it rebooted it got
stuck at "Starting udev" if I remove thew tdm400 it boots OK, but no zaptel
has anyone seen this , and can offer any advice?
Thanks Robb
2008 Nov 20
2
ISDN Cause codes
Hi All
Just been looking at stats for one of my sites, and I'm conserned about
the number of error cause codes being returned from the telco
for example
12000 calls processed
131 are cause code 31* normal. unspecified.*
139 are cause code 28 * invalid number format (address incomplete).*
112 are cause code 1 *Unallocated (unassigned) number.
*this adds up to about 3% of calls not
2004 May 12
1
CISCO 30 VIP phone / 12 SP+ Connection does not free up
Hi,
I am using a 30 VIP phone and a 12 SP+ phone with
Asterisk. When I complete a call outside through the
ZAP device, the phone does not go back to dial tone,
even after I hang up. The line gets disconnected as
per Asterisk console. But the phone stays in the same
state like it is connected. The ZAP line is freed up
and I could make calls from other phones. Only this
phone just remained in
2003 Sep 19
2
Recall doesn't seem to work
Hi
I'm having a problem where the recall button doesn't work
If i press recall before I dial numbers it disconnects me which is what
I would expect, but during a conversation if I want to transfer the TDM
400 just ignores the recall
Any advice would be gratefully received
Thanks
Robb
2004 May 13
3
EXT3 performance on Large (multi-TeraByte) RAID
Has anyone experienced a significant degradation in ext3 performance when using it on a Multi-TeraByte RAID? As part of an experimental setup, I hooked up three 300GB drives and made an EXT3 RAID5 out of them, using the entire space one each drive, and started throwing a large number of files in the size-range 3KB to 50 KB. Then, I deleted the raid, and created a new one, but this time, I used
2004 Jun 25
1
chan_sip.c max number of retries
Has anyone who's gotten this message managed to figure it out and fix it? I've been scouring the mailing list for clues but I'm still no closer.
I have 2 asterisk servers, old and new. I'm trying to switch to the new server. I am using a 2.4 kernel on the new and a 2.6 on the old. I am running 0.9.0 on the old and my sip phones work fine, on the new Ihave tried 0.9.0,
2007 Apr 29
1
Which version of DirectX 9 does Wine support?
Hi,
I came across a game demo, Shrek the third.
The installation went almost smoothly, but the
installation directory it went into was
C:\Program Files\My Product Name
instead of the suggested default:
C:\Program Files\SHReK the THiRD
The demo said it requires DirectX 9c and indeed
it doesn't run because it needs d3dx9_32.dll.
It seems Wine doesn't provide this DLL currently.
Best
2003 Oct 15
2
Odd ringing conditions
I have two questions about incomming ring and extension ringing
1) When an incoming call is detected by asterisk it takes 2 or three
rings before the internal phone ring does anyone know how I can fix this?
2) All internal phone ring on an incoming pstn call but after the call
is answer all the other phone ring for a couple of tinkles how can I
stop this from happening?
Thanks for your
2009 Aug 07
1
Linksys SPA922
Nearly got an SPA922 phone working behind a NAT,
the phone registers, and I can dial out and have two way speech,
on an incoming call the SPA922 rings
I answer and the SPA922 shows "Anwsering" but never does and the far end
continues ringing until the voicemail answers,
this then show as a disconnected call on the SPA922
I'm on the lastest firmware 6.1.5(a)
Thanks in advance
2003 May 13
1
invalid argument 22 when modprobe wcfxs and wcfxo
Hi all
I ahve been having problems loading the wcfss and wcfxo drivers
I get an error message invalid argument and something about post install insmod
failed
the currently load modules do show the drivers loaded but asteris won't start
lsmod
root@slackware:~# lsmod
Module Size Used by Not tainted
soundcore 3332 0 (autoclean)
wcfxo