Dnia środa, 18 sierpnia 2004 16:56, Walt Reed napisał:
> OK, I'm going nuts here trying to correctly identify null values,
> specifically when callerID info is not available.
>
> FYI, I'm running Asterisk CVS-HEAD-08/17/04-13:08:53, and Bison 1.875a
> (debian Sid).
>
> A snippit of my dialplan look
On Wed, 4 Feb 2004 11:13:52 +0200, Dan wrote
> Hi,
>
> From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
>
> > >>I have 4569 opened and forwarded/NATed to my *. I am on the same
network
> > >>as the * server, a friend is remote. After about a minute you
loose the
> > >>connection.
> >
> > > This is
On Tue, 3 Feb 2004 11:33:24 -0600, David Gomillion wrote
> Steven Critchfield wrote:
> > On Tue, 2004-02-03 at 09:53, David Gomillion wrote:
> >> Steven Critchfield wrote:
> >>> On Tue, 2004-02-03 at 06:59, Christopher Lee wrote:
[flames, non-flames etc. snipped]
> what about something like this?
On Sun, 1 Feb 2004 08:21:55 -0600, Rich Adamson wrote
[long snip]
> No, the manual is very verbose but no * examples at all. The
> box sells as either a 323 or sip, with different images
> (sort of like C7960's) and different manuals.
>
> The box does not support the "register" function in ei
On Sat, 24 Jan 2004 10:56:59 -0800, Chris Wilson wrote
> Has anyone had this problem:
>
> (When calling to ext. 1010)
>
> Jan 24 10:50:27 WARNING[-1252262992]: file.c:446
> ast_openstream: File digits/" does not exist in any format
> Jan 24 10:50:27 WARNING[-1252262992]: file.c:734
> as
On Thu, 11 Dec 2003 23:46:46 -0800, andrewg wrote
> Implementation wise, it would be more "frustrating" to kick
> the already registered user off, and make it more likely
> it'd be noticed if there where two registered people.
>
Hmm, frustrating but maybe useful, if you were for example on a
d
On Tue, 9 Dec 2003 08:48:43 +0100, Nicolas Bougues wrote
> On Tue, Dec 09, 2003 at 08:28:27AM +0100, Florian Overkamp wrote:
> >
> > Registration cascading is not possible (I think) but could it be
solved
> > with a shared dial route:
> >
> > Instead of DIAL(IAX/sip.isp.com) could you not
> > DIAL
. I cannot
record audio, playback works OK. Alsa (http://www.alsa-project.org/)
works fine from command line and from my programs, gnophone bitches
about "No input space". After removing the check, works great.
Disclaimer: didn't try with chan_oss
BTW, with ALSA, you might be ev
rked great together.
actually now I'm using alsa (snd-ens1371 and snd-pcm-oss in
/etc/modules) due to some weird behaviour under oss.
hth,
grzegorz nosek
On Thu, 4 Dec 2003 08:44:06 -0400, Carling R. Messina wrote
> Download the latest alsa drivers from sourceforge and make install.
>
On Wed, 3 Dec 2003 20:45:21 +0200, Dan wrote
> Hi,
>
> - Original Message -
> From: "Grzegorz Nosek" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Sent: Wednesday, December 03, 2003 5:08 PM
> Subject: Re: [Asterisk-Users] Iax Client Library Iss
e able
> to receive incoming calls on either iaxcomm or DIAX. Also
> there is a mailing list for the iaxclient library. It's
> [EMAIL PROTECTED] Hope this helps. AJ
>
or:
exten=>1500,1,Dial(IAX/client&IAX2/client,30)
my 0.02pln
grzegorz nosek
On Mon, 01 Dec 2003 15:14:06 +0100, Michael Bielicki wrote
> Low, Adam wrote:
>
> >Second that !
> >
> >-Original Message-
> >From: Cees de Groot [mailto:[EMAIL PROTECTED]
> >Sent: Monday, December 01, 2003 2:35 PM
> >To: [EMAIL PROTECTED]
> >Subject: [Asterisk-Users] Re: Asterisk European
t;pstn gateway, mail server, nfs (/home) and smb file server and
php+mysql app server for about a dozen clients. It worked for quite a
time on a p2/400 but we're currently moving it to a celeron/1700 as
the db is getting bigger.
No asterisk related problems whatever, except for dead
; lines - I don't own any Digium hardware
though :( ) and replace Zap/1/ with Zap/g1/
Works for me with 2x Fritz PCI cards (i4l, as it mostly works and if
it ain't broken, don't fix it.. capi is on my schedule though..)
HTH,
Grzegorz Nosek
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longer run, consider adding a command line option (or even
better, a GUI config item), specifying an int passed to iax_init() in
pc_init() in phonecore.c in gnophone source (I went through the code
so much I almost know it by heart ;). Currently it is passed a zero
which means "the def
On Thu, 20 Nov 2003 14:54:15 -, Linus Surguy wrote
> > So far it seems like the proposed candidates for new lists are:
> >
> > asterisk-newbies (perhaps a better word?)
>
> Maybe asterisk-install ?
>
asterisk-starters ?
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On Thu, 20 Nov 2003 00:47:08 -0500, Dorian Gray wrote
> I yammered:
> > of public resources such as this list. put that FAQ in the list
> > subscribe welcome message or the list sig or the asterisk README or
> > handbook or all of the above...
>
> er, in case it wasn't obvious: s/that FAQ/a link t
r 8000 1.gsm resample -ql
(mindlessly copying the -ql, i don't really remember what it does ;)
hth,
grzegorz nosek
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farm,
does it?
>
> What am I missing? I see many people much smarter than I am
> excited about this, so I am sure I simply failed to consider
> how it will revolutionize everything.
Not that it'll revolutionize anything, it's simply opening another
(however niche) ma
sure i could get it below 100mb or even smaller.
another possibility if you come from the redhat background would be to
use [ducks] mandrake, i remember it had (around 8.2 at least) a
minimal install which amounted to ~60-~70mb (don't know if redhat has
such an option too)
regards
grzegorz
On Wed, 15 Oct 2003 13:49:23 -0500 (CDT), Dave Weis wrote
> On Wed, 15 Oct 2003, M.A. Ali wrote:
> > I am kind of new to asterisk. Here is a little prolem that I am
facing.
> > Here is my problem and questions: I am just adding two gnophone
users to
> > my dialplan, all three systems are within lan
On Fri, 19 Sep 2003 18:10:37 +0100, Scott Stingel wrote
> Have you tried starting asterisk with -c? It should
> give you some detail as to what is happening with the call.
>
> Scott M. Stingel
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On B
e
generated info ("you've been on hold for $time. if you're pissed off
already, dial $phone and complain" ;))
* well, the top of my head seems to end here but i'm sure you'll find
more creative uses :)
cheers,
grzegorz nosek
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i'd:
* download a fresh vanilla 2.4.22 kernel
* untar/bz2 it in /usr/src
* make a link from linux-2.4.22 to linux
* d/l and install openwall maybe? :)
* make menuconfig &c.
* install the kernel (remember lilo.conf & lilo if you use it!)
* reboot to the new kernel
* do wha
he driver - good or bad?), let me know,
i'll dig them up.
>
> Thanks in advance,
> Bryan.
>
> Bryan Nolen
> Lead Developer
> http://Arc.Net.AU
> http://cdonline.com.au
hth,
grzegorz nosek
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ink to /usr/src/linux probably.
btw, make sure you're running the kernel you're compiling the driver for.
hth,
grzegorz nosek
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On Tue, 9 Sep 2003 09:15:32 +0100, Skuse, Phil wrote
> What's the secret to getting sound through Xlite? The SIP
> messages all look OK to me, but the sound isn't coming through.
>
> It was trying to use GSM, so I searched the archive and tried:
>
> disallow=gsm
> allow=ulaw
>
> Now it says tha
sign that came up was a bigger (say, 12/24 ports)
gateway with some embedded Linux running on an industrial PC (as
beefy as circumstances require - any comments?) with plain RJ11
sockets on one side and Ethernet on the other. What do you think
about this?
H
e)
So far, no error messages of any kind, but chan_capi says that CAPI is
not installed. My /etc/asterisk/capi.conf is empty (chan_capi demanded
it and I didn't know what to put there ;)
;)
Thanks in advance
Grzegorz Nosek
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Aste
er YY (the mapping can be static if that
would be a problem)? the connections would be between gnophone and
pstn (i4l driver) and the eavesdropping client would most probably be
x-lite if that matters. i need it as the supervisor needs to control
and instruct newbie workers ;)
tia
grzegorz nosek
> > > No analog modems.
> > >
> > > If your ISDN adapter is supported properly, you can place that straight
> > > into a asterisk box. The analog line would need a X100P.
> > so to sum things up: one x100p for every analog line and one isdn4linux
> > adapter for every isdn line, right? i'd like t
> Subject: Re: [Asterisk-Users] low-cost * (newbie question)
> From: Steven Critchfield <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Organization:
> Date: 01 Apr 2003 09:59:51 -0600
> Reply-To: [EMAIL PROTECTED]
>
> On Tue, 2003-04-01 at 09:31, Grzegorz Nosek w
plain standard
analog internal modems? how? we really cannot afford much funky hardware
(several x100p's, e100p's or an e400p is sort of out of question).
thanks in advance
grzegorz nosek
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