When I set up my Asterisk box at home I didn't want to have to dial 9
to dial off premises, so I gave all my local phones three digit
extensions with this format: 1[1,0]*. My thought is that there are no
area codes that start with 0 or 1, so if I use those numbers, I can
create 20 local extensions
On 8/2/06, Dean @ INKnBITs <[EMAIL PROTECTED]> wrote:
I have been looking through all the web sites about echo problems and how to
solve them on the spa-3000, but I still have not managed to fix mine! I'm in
the UK and have setup all the tones, port impedance to 370+620||310nF, had
the echo Canc
I have an old broken laptop that I am attempting to turn into a mostly
solid state Asterisk box, mainly because I am worried about the fans
or hard drive in my other Asterisk box failing (I already lost one
power supply). The laptop has no screen or hard drive. But it does
have USB and a bootabl
On 3/23/06, Ronald Lewis <[EMAIL PROTECTED]> wrote:
>
> After months of BroadVoice ignoring my trouble tickets for dropped calls,
> delayed termination, etc., I'm throwing in the towel. While they have
> credited $19.95 to my account, they refuse to credit anything more, despite
> ALL of the proble
On 3/13/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
> > The problem with that one is that (as near as I can tell from the
> > photos and the manual) it has no visual MWI.
>
> Is your ATA going to be able to turn on the MWI light on an analog
> phone? How does that work? Just curious.
Yes, I th
On 3/13/06, Kerry Garrison <[EMAIL PROTECTED]> wrote:
> You really aren't going to find an analog phone that works as well as a SIP
> phone for what you are trying to do. Some people suggested the GXP2000 for
> $85 which works ok in a home environment. It is not a top quality phone but
> it has al
On 3/12/06, Martin Joseph <[EMAIL PROTECTED]> wrote:
> > But what the OP wanted was a sulotion that together with the SAP3000
> > makes for something that works even when there is a blackout, since
> > the SPA3000 allows for failover to the FXS port from the FXO port
> > if/when there is no power
On 3/10/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> David - I am not aware of any analog phones that are PoE compatible. Nortel
> and Avaya phones receive line power, but I do not believe they are IEEE
> 802.3af compatible.
It isn't PoE that I am looking for. I want the phone to run off
regul
On 3/10/06, Cory Andrews <[EMAIL PROTECTED]> wrote:
> Dave - I would recommend Aastra analog phones. www.aastra.com The PT-392
> is a real nice model, and inexpensive. These are some of the nicest analog
> phones around for the price.
That looks like a nice phone. But it appears to require an
I am looking for a good analog desktop phone to use with asterisk and
my sipura ATAs. I know I want Caller ID, MWI, a few programmable
buttons (for asterisk features), and no external power supply (so my
users can dial 911 through the SPA-3000 when the power is out). I
spent some time looking at
On 2/16/06, Brent Torrenga <[EMAIL PROTECTED]> wrote:
> Well, I thought and hoped my issue of random hangups on our TDM400P were
> related to busydetect=yes in zapata.conf. The behavior of a call being
> hungup has not changed, however, since setting the busydetect option to
> 'no'. Again, the onl
I would love to run Asterisk on an old laptop, in a mostly solid state
configuration, with no HD. The laptop is slow (Pentium 233), and I
need PCMCIA support (for my network card). Are any of you aware of a
live CD that might work?
Thanks,
Dave
___
-
On 1/26/06, C F <[EMAIL PROTECTED]> wrote:
> OK, some update on this. It's not related to the Sipuras (actualy the
> sipuras are very good at this, since they will re-ring your call). I
> changed my setup to a mediatrix 1204 and I still have the problem.
> Right now I'm looking at:
> 1. Changing t
On 1/26/06, Tomislav Parcina <[EMAIL PROTECTED]> wrote:
Hi Tomislav,
> > I am not very satisfied with this, though. I want to use some
> > features (like Park) that apparently don't work well with reinvites.
> > Have any of the rest of you had any luck troubleshooting this problem?
>
> Your RTP
On 1/16/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote:
> > > I'm using Sipura 3000 as well, however I will have to wait until
> > > Monday about the Switch I'm not sure. So far it looks like Sipura is
> > > at fault. In the mean time I would like to hear
On 1/15/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote:
> > I'm using Sipura 3000 as well, however I will have to wait until
> > Monday about the Switch I'm not sure. So far it looks like Sipura is
> > at fault. In the mean time I would like to hear from others usi
On 1/14/06, C F <[EMAIL PROTECTED]> wrote:
> I'm using Sipura 3000 as well, however I will have to wait until
> Monday about the Switch I'm not sure. So far it looks like Sipura is
> at fault. In the mean time I would like to hear from others using the
> Sipura 3000 FXO if they have the same probl
On 1/13/06, C F <[EMAIL PROTECTED]> wrote:
> OK, I'm using it with Sipura, Polycom, and Asterisk, and only when
> Asterisk is in the media path, do I have this problem. Also I haven't
> noticed this problem with Polycoms alone. So I'm assuming it's a
> Sipura problem. But I might be wrong.
That m
On 1/13/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote:
> > 2. Is asterisk in the media path?
>
> I don't think I understand this question.
I learned a little about this. Assuming I understand this correctly,
I think asterisk is in the media path. canreinvite=no for all
On 1/13/06, C F <[EMAIL PROTECTED]> wrote:
> I have this exact same problem, so far I have not been able to figure
> out whats going on. I also get the follwoing errors:
> ==
> Didn't get a frame from channel: SIP/119-41c6
> Bridge stops bridging channels SIP/119-41c6 and SIP/101-264e
> ==
34506
I can't tell whether it is a configuration problem or a bug.
Have any of you run into this problem before?
Thanks,
Dave
On 1/12/06, Thczv F. Thczv <[EMAIL PROTECTED]> wrote:
> I am hoping some of you can help me troubleshoot this problem I am
> having with my home
I am hoping some of you can help me troubleshoot this problem I am
having with my home asterisk machine. I have incoming POTS service
using a SPA-3000 (extension 119). Calls on that line go to an
attendant recording that offers a menu choice: press 1 for Nancy,
press 2 for the rest of us. In rea
I work for a telecom company that allows me to peer my Asterisk box to
their system for free. Pretty neat. I have everything working except
that I can't get inbound VoIP calls using the DID number that my
company assigned for me. Today, I finally discovered the source of
the problem: For various
Hello all,
I have [EMAIL PROTECTED] working pretty well, and have been trying to fix
a few things. Today I tried to make Asterisk send voicemails to my
e-mail account. I installed Webmin, which worked fine, and tried a few
things that I read on line, mainly involving the sendmail application
and f
Hello all:
For fun, I am learning about Asterisk, and trying to get Asterisk
working at my house. I installed [EMAIL PROTECTED] It seems to be
functioning fine. I installed a couple of softphones, and have them
registered with Asterisk. I actually work for a CLEC, and I have
registered my Aste
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