Robert Hajime Lanning wrote:
Can you show me an ad for an IP phone which doesn't say it includes an
echo canceller? A real phone, I mean. Not some thrown together half
baked softphone, many of which do a very poor job.
I haven't once talked about soft phones. I don't use them. I am
talki
On 14:10, Fri 11 Feb 05, Remco Barende wrote:
> Hi list!
>
> I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The
> instability is driving me crazy however.
>
> I'm having continuous problems where inbound calls will not work after
> some time of operation (the number then a
On 00:07, Fri 11 Feb 05, Robert Rozman wrote:
> Hi,
>
> Covide looks interesting. Is this a killer combination of groupware and
> Asterisk I was looking for ?
> Is it open source ? Do you have any more english info ?
>
> Thanks in advance,
>
> regards,
>
> Rob.
Hi,
We believe it's the kille
> Can you show me an ad for an IP phone which doesn't say it includes an
> echo canceller? A real phone, I mean. Not some thrown together half
> baked softphone, many of which do a very poor job.
I haven't once talked about soft phones. I don't use them. I am
talking about hardphones that talk
> The sidetone is 'always' generated within analog and digital phones.
> It never comes from any source outside the phone. In analog phones,
> it derived from the hybrid within the phone. On digital phones, its
> basically firmware.
I never said that sidetone was generated outside the phone.
Th
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Hello everyone,
I've got an old AVM all-in-one ISDN box lying around. Currently,
it's attached to my server via USB and CAPI4Linux and additionally
has an analog DECT phone attached to one of its TAE ports.
As I'm planning to switch to VOIP via cable
Nitesh Divecha wrote:
Hey All,
Just finished installing Asterisk and configured all the necessary
parameters to start.
I can’t seem to find the Meetme application in my asterisk directory.
I downloaded asterisk from CVS and installed it and all my Snom phones
are working and voicemail too.
Robert Hajime Lanning wrote:
Wrong. Look at any cellular phone or IP phone. They all have echo
cancellers. If you switch these cancellers off the results are
generally bad. What they need to remove is the acoustic spill
from the earpiece to the mike. This can be a surprisingly strong
signal.
> Wrong. Look at any cellular phone or IP phone. They all have echo
> cancellers. If you switch these cancellers off the results are
> generally bad. What they need to remove is the acoustic spill
> from the earpiece to the mike. This can be a surprisingly strong
> signal.
While acoustic "spill"
GFI MailSecurity's HTML threat engine found HTML scripts in this email and has disabled them.
From couple of weeks i am working on asterisk fax but was
not successful.
I am able to receive only half fax documents and its ending with blurred lines.
I have tried with almost all the versions of
I have a SNOM 190 phone at home. It works well.
On the display it shows the time, 3 soft key labels and also the number of
missed calls.
If I see a missed call I can use the CallLog soft key to see a list of calls
that are divided into:- Missed, Received and Dialled
I have extensions.conf set up
Eric Bishop wrote:
Just out of interest,
When echo occurs (the type where I hear myself echoing as I talk) what
is bouncing against. Is it the other caller's equipment, the central
office or something in between?
With 2-wire analogue line you will have echo from the hybrid at the
local exchan
Robert Hajime Lanning wrote:
Just out of interest,
When echo occurs (the type where I hear myself echoing as I talk) what
is bouncing against. Is it the other caller's equipment, the central
office or something in between?
When you are talking via 4 wire or VoIP phones there is a seperate
queues.conf
; Persistent Members
;Store each dynamic agent in each queue in the astdb so that
;when asterisk is restarted, each agent will be automatically
;readded into their recorded queues. Default is 'yes'.
;
persistentmembers = yes
Julian
Robert Rozman wrote:
Hi,
I noticed that age
Hi,
I am a newbie in asterisk. trying to configure firefly third party edition
to connect to aserisk 1.0.3 im able to authenticate but cannot dial
extensions. I have been reading the documentation cant seem to find the
correct configs. Attached the error message and configs. What am I
missing?
*
> > When echo occurs (the type where I hear myself echoing as I talk) what
> > is bouncing against. Is it the other caller's equipment, the central
> > office or something in between?
>
> When you are talking via 4 wire or VoIP phones there is a seperate
> outbound audio channel and inbound audio
> >>The bottom line for those asterisk readers that have actually read this
> >>far is to use complex & lenthy passwords where possible, and some sort of
> >>alerting mechansim when xx number of passwords are guessed incorrectly
> >>(such as an account lockout mechanism with alerts as just one of m
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