Jesus Mogollon wrote:
Hi all
Does anyone know of any motherboards with PCI slots that can take
the TE412P card? Is there such a MB for Athlon 64 or P4 procs?
I have a TE410P working with an ASUS P5MT mobo with Intel Pentium D
processor.
___
-
Hi,
Ive been playing on a asterisk to orion gsm box E1 pri setup.
I have achieved incoming calls to be passed to my asterisk box
successfully but outgoing calls will just
I have tried playing with various pridialplan and overlapdial settings
and with no success. If anyone can make more sense f
> "PJ" == Pavel Jezek <[EMAIL PROTECTED]> writes:
PJ> tunneling small rtp packets through vpn has big overhead, better
PJ> to use application level encryption - encrypted iax or srtp.
IPSEC in transport mode without NAT has a very low overhead.
/Benny
_
If you are using one line for fax only then you do not need to do fax
detect. Put it in its own context and make ths s extension be Rxfax.
Everyone always tries to over-compliate stuff, and it seems to me the less
you know about asterisk the more elaborate and overcomplicated schemes
people can c
In Your Setup The First Asterisk Box Should Not Display This Music On
Hold Notice Unless It Is Playing It To The Other Asterisk. In Other
Words It Seems That You Are Putting A Channel That Hears Music On Hold
And Playing To It Music On Hold
On 12/15/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
On Fri, Dec 15, 2006 at 06:32:19PM -0800, Yuan LIU wrote:
> When booting Ubuntu 6.06.1 (Linux 2.6.15-27-386), wcfxo would load but not
> configure. I have three ways to manually force wcfxo to configure: 1)
> ztcfg, 2) modprobe -f wcfxo, or of course 3) unload and reload wcfxo. Each
> works eq
On Fri, Dec 15, 2006 at 06:41:58PM -0800, Yuan LIU wrote:
> Hardware is an SM56 card (X100P clone). When the line hangs up, ztmonitor
> displays full bar (or whatever maximum allowed by rxgain) in RX. It only
> drops zero when the line picks up (and remote was silent). Is this
> something of
I havent tried this setiing yet but i dont think separating ports by , will
make asterisk listen on multiple ports ( i read it somewhere but cant
remember ) . I will try it out sometime .
On 17/12/06, EWV2 <[EMAIL PROTECTED]> wrote:
Bindport is for [General]
Port is for [Device]
You con
This went little bit offtopic :P but i dont mind . Here's how i solved it (
just if some one in future goes in this problem and end up here by googling
) .
Port 5060 was blocked by isp incoming as well as outgoing for few
extension's isp . On server i did
iptables -t nat -A PREROUTING -i eth0 -p
Bindport is for [General]
Port is for [Device]
You con use multiple ports in this format port=5060,5061,5091 or
bindport=5060,5061,5091
That's what I have read
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sales
Sent: Saturday, December 16, 2006 2:00
Hi,
I noticed that the sound directory is missing from asterisk-1.4.0-beta4.tar.gz.
This directory (7 M) witch existed in asterisk-1.4.0-beta3.tar.gz has GSM Core
Sounds and some MOH.
Does anyone know why it has been removed from the latest beta?
Regards.
_
Then there is no way to make asterisk listen on multiple port ? Currently
iptables 5091 forward to 5060 is working but this should really be a
asterisk feature :(
On 16/12/06, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
Mail list wrote:
> Yes i read that on voip-info wiki but i have bind
If you are really new to linux then go for trixbox . I started with trixbox
and eventually went away from it by removing extra stuff and putting custom
compiled asterisk's and removing their rpm's . If you are good at linux then
definitely go for debian + asterisk or centos+asterisk and put free
Noah Miller wrote:
Last week I asked about MWI indicators on wireless phones that would work
with Asterisk. I sent a message off to Panasonic asking them about it
because in their ads they specifically stated that the indicator works
with and requires phone company voicemail subscription.
> That
On Saturday 16 December 2006 5:14 am, Phil Finkler wrote:
> Hey all,
>
>
>
> I've been doing a lot of playing, and a lot of reading, and it seems
> people are split as to whereas if they're running their favorite Linux
> distro and asterisk or Trixbox. I'm getting closer to really looking at
> a p
Hello,
I have a rather odd problem with Asterisk detecting faxes. I have two
POTS lines coming into the box (TDM400P). Line 1 is for voice, Line 2
is fof fax. When I set them up with channel => 1-2 in zapata.conf,
all is fine, but as soon as I have two channel => definitions,
Asterisk is
On Fri, 15 Dec 2006, John Novack wrote:
> Google is your friend!!
>
> http://www.eweek.com/article2/0,1895,1773983,00.asp
Which discusses the Vonage case, which was settled, and says
"A larger ongoing question is simply how VOIP will be viewed by the FCC, a
political organization where a major
I am using asterisk 1.2.13 with multiple SIP service providers to handle
my out going call. I have my dialplan list below.
It works fine with one exception -
when my primary SIP provider off air - not responding to the SIP INVITE-
it took 60 seconds for sip.channel to return with DIALSTATUS=NOANS
I forgot to mention this is an x86_64 Pentium-D system.
Regards,
David
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1.4 beta-3 compiled fine under Fedora 6 w/IMAP storage. I get this on beta-4.
Any Ideas?
[CC] app_voicemail.c -> app_voicemail.o
In file included from /usr/src/imap-2006c1/c-client/osdep.h:63,
from /usr/src/imap-2006c1/c-client/c-client.h:42,
from app_voicemail.c:65
Well it is not clear - their ads say one thing and they say another. At
best there is confusion. If you really like the phone you could buy it
somewhere where you could take it back and check it when you get it.
Doug
On Sat, 16 Dec 2006, Noah Miller wrote:
> > Last week I asked about MWI indicat
Mail list wrote:
Yes i read that on voip-info wiki but i have bindport = under device
(extension) which should make that extension work on other port but its
not working . :(
No, bindport= under the device section is ignored because it is not
supported.
_
Can anyone help with the compile error? 1.4 beta-3 compiled fine under
Fedora 6 w/IMAP storage. I get this on beta-4.
[CC] app_voicemail.c -> app_voicemail.o
In file included from /usr/src/imap-2006c1/c-client/osdep.h:63,
from /usr/src/imap-2006c1/c-client/c-client.h:42,
I've been doing a lot of playing, and a lot of reading, and it seems people
are split as to whereas if they're running their favorite Linux distro and
asterisk or Trixbox. I'm getting closer to really looking at a production
environment and I'm just looking for any opinions. I'm really enjoying
On 12/16/06, Phil Finkler <[EMAIL PROTECTED]> wrote:
Hey all,
I've been doing a lot of playing, and a lot of reading, and it seems
people are split as to whereas if they're running their favorite Linux
distro and asterisk or Trixbox. I'm getting closer to really looking at a
production enviro
Hey all,
I've been doing a lot of playing, and a lot of reading, and it seems
people are split as to whereas if they're running their favorite Linux
distro and asterisk or Trixbox. I'm getting closer to really looking at
a production environment and I'm just looking for any opinions. I'm
real
It is scheduled for 9 januari. (If you ask nicely on
[EMAIL PROTECTED] and promise to give good feedback, you might be
able to get a beta version earlier ;)
Zoa
Vicky wrote:
I have configure it by using the *2 atxfer feature of asterisk but its
not as good as other attended transfer which s
I have configure it by using the *2 atxfer feature of asterisk but its not
as good as other attended transfer which sipphones give ( like sjphone where
you can switch between two anytime ) . Also tried zoiper but it do not have
even blind transfer yet . Any idea when idefisk 2.0 is going to be rel
Idefisk 2.0 will have it.
Zoa
Mail list wrote:
Is there any good iax2 softphone capable of attended transfer ( like
sjphone for sip ) . ? I tried iaxcomm and idefisk both seems unable to
handle attended transfers.
_
I haven't tracked this down to anything on my system yet, but has
anybody else upgraded to 1.4.0b4 (from 1.4.0b2) and found that asterisk
core-dumps on startup?
The last few lines in messages before dump are:
[Dec 16 10:44:03] WARNING[7958] translate.c: plc_samples 160 format 6
[Dec 16 10:44:0
Last week I asked about MWI indicators on wireless phones that would work
with Asterisk. I sent a message off to Panasonic asking them about it
because in their ads they specifically stated that the indicator works
with and requires phone company voicemail subscription.
> That indicator will not
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