Last week I got an email from Link2VoIP saying that they are shutting it
down in a few months. Sighting competition and the unethical changes being
made to the Internet by special interest groups.
I use Link2VoIP for termination, connecting my Asterisk servers to the
regular old telephone company.
I have an Asterisk server it runs great with SIP phones, soft SIP phones
(twinkle) and a soft SIP phone app on my Android phone but I am having
problems getting two ATA boxes working. I have a Linksys PAP2T, it is
unlocked and I have used them before with no problems. I was able to
receive calls wi
> > >
> > > Loud scrathing sound? sometimes a card problem, try on other hardware.
> > >
> > > Pci interrupts, also maybe sync problem (you can enable b410 clock in
> > > misdn-init.conf)
> > >
> > >
> > > Also turn off a
Not sure what you are asking for here? It looks like you have a config file
that sets a clock speed but I do not know what file that is.
How can I find this informaiton?
On Sat, Mar 8, 2008 at 5:48 PM, Fons van der Beek <[EMAIL PROTECTED]>
wrote:
> what clock?
> rxclock
> crystalclock
>
>
> I cu
t, Mar 8, 2008 at 10:24 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> > IRQ's seem to have been the problem. Thanks Steve Totaro for that tip.
> >
> > The Digium cards were at the same IRQ as the IDE controller, I moved the
> > cards and hard drives to
s bad. ALWAYS use surge suppression on your lines!
>
> Thanks,
> Steve Totaro
>
> On Wed, Feb 27, 2008 at 11:36 AM, Royce Souther <[EMAIL PROTECTED]> wrote:
> > I have setup a few Asterisk systems for customers using Digium TDM400
> cards
> > and Aastra phones. No prob
I have setup a few Asterisk systems for customers using Digium TDM400 cards
and Aastra phones. No problems with sound quality at all except at this one
site.
Every time I try their system I don't hear any problems but they tell me
that it is really bad. They describe it a a loud scratching sound.
Are you in the Southern Alberta area? I am putting on a free VoIP * workshop
on Friday afternoon. Everyone is welcome to attend.
This is to introduce local business to the benefits of VoIP using Asterisk.
If you want to attend or if you have clients you think could benefit from
this please email
, all indicators are that this was the solution I
needed.
On Sun, Feb 10, 2008 at 12:33 PM, Royce Souther <[EMAIL PROTECTED]> wrote:
> I have a network of offices using Asterisk that are connected via IAX2
> trunks. The trunks work great for a day or two then for no reason at all one
> e
I have a network of offices using Asterisk that are connected via IAX2
trunks. The trunks work great for a day or two then for no reason at all one
end of the trunk will become UNREACHABLE while the other end is still
connected. The only way to fix the problem is to shutdown Asterisk completly
then
I am using Asterisk 1.2.18 with FreePBX 2.2.0.
I have two Asterisk systems with an IAX2 trunk between them. I want to make
each end so when a user dials the local 7 digit number for the other end it
will try to rute the call through the IAX2 trunk before trying the PSTN
lines. When the call comes
I am trying to setup SIP to SIP calling between Asterisk managed networks. I
want to make it so that people can call SIP:[EMAIL PROTECTED] and they
connect to my Asterisk and get my external IVR then they can dial my
extension or navigate extensions just like they would if they had called
using a P
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