Start Asterisk as "asterisk -cvvv"
jazzy singh wrote:
> I did that but it still is dyingjust to let you guys know I'm using
> lumenvox as well, when I restart asterisk it restarts and then it just dies
> right away without any error message. Please help
> Thanks
>
>
>
>
> Date: Thu, 2
William Moore wrote:
>> Dial() already supports dialing multiple lines at once and connecting the
>> first answered line to the user. I want that exact same functionality but
>> would like to be able to space out (or stagger) the dialing of each line.
>> An example might be I want to dial 5 phone
Russell Bryant wrote:
Eric "ManxPower" Wieling wrote:
I discovered today that in -trunk all the docs have been converted to
TeX format. Are these docs the only ones that will be in the release
tarball? If so, I think it is a bad idea.
We do not want to make people install y
Andrew Kohlsmith wrote:
On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote:
chan_pan :-)
PAN/nokia
PAN/peter
PAN = Personal Area Network
Eww... this isn't using the PAN profile at all, so I don't think that'd be
right... chan_HFP or HSP would be my guess if you wanted to go that route.
Klaus Darilion wrote:
Vazir wrote:
How than my CISCO AS5350 gets H323 VOIP and echo-cancels so exelently
:))
If it is a plain H323-H323 call then there should not be any echo at all.
If you have a PSTN<->VoIP call and the PSTN leg uses analog lines then
the PSTN side will create an ech
Tzafrir Cohen wrote:
On Mon, May 07, 2007 at 02:45:48AM -0500, Eric ManxPower Wieling wrote:
syd wonder wrote:
Hi all. I'm trying to hook into a chan_zap.c function that enables echo
cancellation. I think I've identified the spots to hook into but am
wondering if someone can an
syd wonder wrote:
Hi all. I'm trying to hook into a chan_zap.c function that enables echo
cancellation. I think I've identified the spots to hook into but am
wondering if someone can answer a few questions or provide some guidance.
1 - I'm using an external device (Sipura SPA3000 FXS/FXO) co
Just how many pickup groups do you need?
If you are assigning one pickup group for each extension then your
design is wrong.
Pavel Jezek wrote:
if this limitation is really true, it is challenge for some rework,
because with 64 pickup groups limit, it's usefull only for small
companies :-(
Roy Sigurd Karlsbakk wrote:
hi all
I've been fighting this for a while, and telco tells me i'm doing it
wrong and all, although I'm doing it after the book (or so I beleive).
To divert a call, I do as follows
exten => s,n,Set(CALLERID(rdnis)=${CALLERID(number)})
exten => s,n,Set(CALLERID(num
No it is not possible.
Alexei Volkov wrote:
Is it possible (in theory) to make asterisk server multiple sip endponts
configured with same sip credentials.
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Why not allow group= or make group 0 mean "no group".
What I do is set channels that I don't want in a group to be group 0 and
never use g0 anywhere
Nic Bellamy wrote:
Olle E Johansson wrote:
5 nov 2006 kl. 16.43 skrev Gil Kloepfer:
I've discovered that in configuration files such as the
Tony Mountifield wrote:
In article <[EMAIL PROTECTED]>,
Eric \"ManxPower\" Wieling <[EMAIL PROTECTED]> wrote:
Alistair Cunningham wrote:
When I was an engineer working mostly with IBM Websphere Voice Response,
and we wanted to take a machine down, we would quiesce the tr
Tzafrir Cohen wrote:
On Mon, Oct 16, 2006 at 06:43:53AM -0500, Rich Adamson wrote:
What I'm worried about is trying to fix it in the current code, since it
will change quite a few things that are needed today,
and break backwards compatibility. I've tried, but failed, in chan_sip2
:-) and that
Peter Beckman wrote:
Hey folks --
Noticed that Asterisk doesn't use libwrap. Any reason? Could it be added?
It would be handy.
Does libwrap support UDP?
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Sounds like ECFO.
Echo Canceler Freak Out, this happens when the rxgain is too high and
the echo canceler freaks out. Some users describe it as "screeching",
"feedback", "static", or other useless terms. If users report "static"
on a system where there cannot be static (all digital, PRI, SIP
/path/to/src/asterisk/doc/README.backtrace
Kohler, Jeffrey wrote:
I was able to eventually figure it out. For anyone as linux unsavvy as
myself:
- # gdb /usr/sbin/asterisk /tmp/core.21713
- # bt
Sorry for the stupid question
From:
SIPura has that feature.
Kevin P. Fleming wrote:
- John Lange <[EMAIL PROTECTED]> wrote:
The question then becomes, if most client devices support keep-alive
is
there still a purpose to having it on the server side as well? How
many
client devices support keep-alive? I know Linksys products
Alejandro Kauffmann wrote:
Eric "ManxPower" Wieling wrote:
My only comment is that using "extension" to refer to an entry in the
current sip.conf is insane. SIP devices are not extensions. They are
devices which may or may not have more than one entry in sip.conf (one
Asterisk assumes all incoming calls will be authenticated. You want all
incoming calls to not require authentication. This is not a common
usage of Asterisk.
There is an option called insecure=very This option is shown in the
/path/to/src/asterisk/configs/sip.conf.sample
I assume the Wiki
Your sip.conf needs to accept calls from anyone.
[EMAIL PROTECTED] wrote:
Hello,
Anybody could explain me why asterisk spend time to
send back to proxy or sip agent authentication
messages 407
nobody can call me from other domains.
can we disable authentication for none peers or users
Aster
Steven Critchfield wrote:
On Fri, 2006-03-03 at 11:20 -0500, Paul wrote:
Steven wrote:
Performance and security is not special to any interface of asterisk.
Performance isn't usually a quality of AGI that is taken really
seriously. Look at the many applications Tilghman has written so that
o
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