segfault
> > when rxfax is called.
> >
> >
> >
> > On 07/12/06, Matt Gibson <[EMAIL PROTECTED]> wrote:
> > > Same thing occuring here, on gentoo as well :(
> > >
> > >
> > > On 07/12/06, Chris Glover <[EMAIL PROTECTED
Hi,
I have installed the latest version of asterisk(1.4.0-beta3), and built
app_rxfax/txfax. I'm using spandsp from here,
http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz
Everything builds ok. I had to manually apply the patch from the site so
configure would spot
The clock on your computer still thinks it's April... Please fix!
As for editing C code.. any decent text editor will do the job.
On Mon, 2005-04-25 at 13:26 +0530, Bharat M. Sarvan wrote:
> Hello Everybody,
>
> I was going thru the C code of Asterisk. Does
> anybody kno
On Fri, 2005-05-13 at 09:02 -0700, trixter http://www.0xdecafbad.com
wrote:
> Does anyone know of a BYOD provider that terminates calls to NCFA
> numbers (UK 'national rate'). I enjoyed broadvoices unlimited to those
> numbers, but this is getting silly now, it doesnt work and no answer if
> after
On Thu, 6 Jan 2005, John Middleton wrote:
Ive just installed a TDM4000P with 4 fxos. The zaptel config is fine,
zttest comes back with configured. If i call a line when zttest it
shows on the display,and then goes when the line drops.
In * when a call comes in, it follows my dialplan and answers th
On Tue, 4 Jan 2005, John Middleton wrote:
Hi,
Anyone used this service, any comments on reliability/support?
Hi John,
I've been using this service for a while now, works very well.
Had a couple of minor problems with IAX originally, but they were sorted
within a couple of hours, and they even call
On Wed, 8 Dec 2004, Michael Welter wrote:
Check the mains supply, there might be something with a big inductive
load, that causes some ripples that are ending up on the DC outputs of
your PCs PSU.
HTH
Chris
This was a situation where I had to recover the root password, so I wasn't
able to ssh i
On Wed, 8 Dec 2004, Michael Welter wrote:
I went on a service call yesterday to find an asterisk system with a T100P
card on a Qwest PRI and a TDM40B card connected to fax machines. The TDM40B
LEDs were not lit, and the system did not respond to keyboard input.
However, calls were being processe
Hi,
I'm thinking about buying a Sun Blade 100 from Ebay. I see that it has PCI
slots. I want to run Gentoo Linux on it and install my X100P card.
My Question is...
Will the X100P card work happily with Linux on a Sparc processor?
Has anyone every tried this or the TDM400 series?
Thanks in Adv
Hi Matt,
You're using the wrong signalling type, should be fxs_ks
HTH
Chris
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Chris
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On Thu, 26 Aug 2004, Matt wrote:
> I've been strugling with this for two days now and I'm making no pr
Hi,
I use Gentoo, and find it's documentation and package management
excellent.
Try here
http://www.gentoo.org/doc/en/index.xml
Chris
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Chris
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On Wed, 28 Jul 2004, usedcanon wrote:
> Pete
Is the path to soxmix in the $PATH environment variable when asterisk
starts. If you're running from an init script it may not have path set at
that point.
When you log in, you set the path variable. Have you tried putting
explicit paths into the command in your extensions.conf?
IE /usr/bin/soxmi
On Thu, 15 Jul 2004 [EMAIL PROTECTED] wrote:
>
> Note "The previous method for terminating IAX2 calls using Connect! will
> cease to be available at midnight (GMT) on August 15th, 2004." The
> message I got was at 1:51 AM EST. That means I was given negative 5
> hours and 51 minutes to make this
Also www.voiptalk.org
They can do SIP and IAX.
HTH
Chris
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On Wed, 14 Jul 2004, Dameon D. Welch-Abernathy wrote:
> On Wed, 2004-07-14 at 11:41, Johannes van Hulst wrote:
> > Can som
Hi,
I have a Cisco 7960 with SIP firmware which works perfectly with
Asterisk. I did get it working using chan_sccp originally. I couldn't make
chan_skinny work. All without callmanager.
HTH
Chris
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Chris
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IAXT
Hi,
I've noticed the same thing with my 7960 running SIP 7.1. I came to the
conclusion that the Cisco does some clever echo cancellation, and that
pause is the echo canceller training. I don't think it's an Asterisk
problem. My Grandstream 101 works perfectly.
Chris
--
Chris
---
On Wed, 23 Jun 2004, Matt wrote:
> >>I've been having similar problems to you. I found after reading an
> unrelated post, about the Jitterbuffer option in
> >>iax.conf, setting this to yes has made things much better.
> Out of interest what have you set for your
> dropcount
> maxjitterbuffer
> Max
Hi,
I've been having similar problems to you. I found after reading an
unrelated post, about the Jitterbuffer option in iax.conf, setting this to
yes has made things much better.
Now the problem I have is that telappliant have lost one of there external
routers so I can't connect at the moment :-
On Sun, 20 Jun 2004, Kevin Walsh wrote:
> It's possible that the time/date is also encoded into the Caller*ID
> signal. I haven't had cause to look into that. It's possible that
> the DECT phones ignore the local time and use the time provided by the
> Sipura (if the Caller*ID signal does indeed
I think you'll find that Telsta is an Australian telco, not Austrian!
Only a few miles out :-)
--
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On Mon, 14 Jun 2004, Wolfgang Pichler wrote:
> sorry, i don't know what you mean with:
Hi,
Just a thought, are you using the XT-PIC mode of IRQ address (old style)
on the new IO-APIC version that Kernel 2.6.x supports?
I found with my Intel 537 modem that it would hang a machine using XT-PIC
(admittedly it was IRQ sharing with all the other crap in the machine) but
using IO-APIC th
> Chris Glover wrote:
>
> > I use voiptalk via my DSL connection. It seems to work very well.
> > Originally I was using the connection in Sip mode, but had problems with
> > DTMF, I could only get it to work on outgoing calls, or incoming if I
> > changed mode, but
Hi Matt,
I use voiptalk via my DSL connection. It seems to work very well.
Originally I was using the connection in Sip mode, but had problems with
DTMF, I could only get it to work on outgoing calls, or incoming if I
changed mode, but not both. I switched to using IAX last week, which they
set up
Hi all,
First an introduction. I've been using Asterisk at home for a few months,
but only just joined the list. So far I'm very impressed.
I'm glad I'm not the only one who's been having the problems Julian has.
This is what I get when I try placing a call into my asterisk system. I'm
using voip
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