That's a problem that needs to be discussed with the provider of those POTS
lines..
With POTS lines, each number has a unique number, but the telco builds a hunt
group for you..
For example, maybe you own phone number 555-1000. Your POTS lines may have
555-1000, 555-1001, etc, all the way to
I think I'm missing the beginning of this thread, but I had this exact problem
with a Call Manager going to two SIP providers, one of which was BW.COM.. I
don't know if it will help, since presumably you're using asterisk, but with
the call manager, the problem was that there was no
This is interesting to me.. I'm a newbie, so please forgive a dumb
question, but what use is it to play a message if you don't pick up the
phone first?? Who's hearing it?
-Original Message-
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs.
Email me off list with the phone part numbers, and I'll see what I can do.. It
probably depends on the level of cisco certification the company has. I dont
know if we can do better, but I'll see!
Steve
[EMAIL PROTECTED]
From: Louis-David Mitterrand
http://www.x100p.com/products_2.htm
Anyone ever use this box? Hows it compare with the
Iaxy? Id like to buy one or the other.. The Iaxy is appealing because
to me, it seems less no name, but this one says that it supports
using hostnames, whereas apparently the iaxy only supports IP
I liked the ringer that read the phone
number too, but a couple months ago, I did a firmware upgrade, and that ringer
option went away Do you have the latest firmware?? I upgraded because
of a problem with my phone losing registration, which is now fixed, but I lost
that really cool
Depends what you want to do!
Do you want to do VoIP over that T1 to a provider or IP telephones?
Do you want to hook up to the PSTN through that T1 as 24 voice channels,
through a T1 card on your asterisk?
If you want to use the T1 as 24 voice channels, the Telco is going to
have to re-provision
. Similarly, I would point my 800 number to a
DiD hosted by a VoIP provider that would then route the call back to
me. If that is an incorrect assumption, please let me know.
Regards,
Warren
Steve Jones wrote:
Depends what you want to do!
Do you want to do VoIP over that T1 to a provider or IP
ringer.
I guess they decided we didn't need that ringer.
Do you update off of their system, or do you have your own tftp server?
On 6/19/06, Steve Jones [EMAIL PROTECTED] wrote:
I liked the ringer that read the phone number too, but a couple months ago, I
did a firmware upgrade
If youre going to have to open
ports on your firewall for SIP anyway, then why not put the server on the
inside? That being said, I dont know if youd need to punch holes
for the phones being trusted and the server on the outside..
Personally I dont like the ideas of
having a server
I would say its only profitable if
youre getting ONE T1 instead of two??
From: Gabriel Afana
[mailto:[EMAIL PROTECTED]
Sent: Monday, June 19, 2006 1:34
PM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How
to use a data T-1?
I went through the same thing on my home system a couple months ago, and
asked similar questions.. The conventional wisdom is that it
depends... It depends on your local loop length, quality, taps on the
line, etc..
It seems that most people who have the sangoma cards with hardware echo
Sounds like something is detecting whatever vocal tone is occurring as a
DTMF sound, and conveying it inband.. I'm not an expert, but what DTMF
settings do you have?
-Original Message-
From: Servetas, Andrew [mailto:[EMAIL PROTECTED]
Sent: Friday, June 16, 2006 1:22 PM
To:
. Guess I
should locate my oldest 7960 first, in case there are sparks and a fire :-)
No, you want the NEWEST one, because its
still under warranty!! ;-)
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users
I was trying to modify some rules in my [EMAIL PROTECTED] config this morning,
and I
was having a LOT of trouble, and not understanding why it was ignoring
some outbound rules, and it wasn't 'till I made all my NXXNXX type
characters all uppercase that it worked properly. I didn't think it
I've got a Motorola vonage box, and I've used it w/ AAH using a digium 4 port
card, and with a clone modem FXO, and both have worked well, except for the
echo that caused me to upgrade to the real digium card...
From: Paul [mailto:[EMAIL PROTECTED]
Sent: Tue
First of all, I assume that since you're asking the question, you want to
trunk, or send/receive calls that are on the OTHER SIDE of a proxy from you.
Certainly asterisk, as a PBX, can service local IP phones, and connect to PSTN
lines, without regard to ANY internet connection.
Proxy servers
Its possible that for security
reasons, it doesnt let you log on remotely with the default passwords. From
the console, change the password to something else unique, and it should work.
You should probably do each of these:
passwd-maint set master
password for web GUI
On mine, I had that happen, until I turned off subscribe to MWI or
something like that in the config (sorry - I can't remember the exact
verbiage.) I also upgraded the firmware around the same time, but I
think from advice I got on this list, the MWI setting was the reason...
-Steve
I didn't get that message, but I got a we're sorry for sending out the
bogus messages message, saying that it was an error..
-Steve
-Original Message-
From: Steve Prior [mailto:[EMAIL PROTECTED]
Sent: Thursday, May 04, 2006 6:40 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Just as another datapoint, I have a cheap (~$17) walmart cordless phone
at home hooked to my digium dual FXS card, and it works great, with the
possible exception that there is a buzz (I perceive it as ground hum)
for about the first 4 seconds of EVERY call, and slowly it diminishes.
I don't know
I can't find a pocket pc version of that on the iaxcomm website.. Only
linux, Mac, Windows.. Can you send a link? This is exactly what I'm
looking for!! Thanks!
-Original Message-
From: Robert Augustyn [mailto:[EMAIL PROTECTED]
Sent: Tuesday, April 25, 2006 12:38 PM
To: 'Asterisk
I had a similar problem with a GS101, although with mine, I could make
OUTBOUND calls from the phone, but because it wasn't registered, it
wouldn't ring if called. I don't know the exact solution, but two
things I did was to tell it NOT to subscribe to MWI in the GS config
itself, and second, I
Why not use [EMAIL PROTECTED] It's got the AMP/FreePBX already installed,
so it'd be easy for them to maintain, and should do what you want..
-Original Message-
From: Nabeel Jafferali [mailto:[EMAIL PROTECTED]
Sent: Thursday, April 20, 2006 2:40 PM
To: asterisk-users@lists.digium.com
First, I'm surprised it didn't detect your network card.. I'd say it'd be
worth putting in another network card. It's autodetected every one I've
used
Second, I am not sure if it has built in functionality to do what you want,
but it certainly will come closer out of the box than a
I had a similar problem, and it was because my hostname had issues... I'm not
sure why/how, but if my hostname was valid, and had a valid fwd/reverse dns
entry, everything was OK again..
-Steve
From: Josué Conti [mailto:[EMAIL PROTECTED]
Sent: Wed 4/19/2006
Im using SixTel as a test (Opened
account w/ $10) and am happy with them so far In their basic
service package, they dont charge a monthly fee, and its outbound
only, and you get charged for every minute. I paid for a DID, which is
$1.50 or so per month, and it lets me receive inbound
I had the same question, and I want to make sure I'm clear. This
implies to me that Asterisk itself doesn't use SMTP, but rather dumps a
message into some directory that Sendmail on the same box will see and
process? I have no problem getting Sendmail to use a smarthost, but am
I
their
board without the echo cancellation. I'm thinking you'd be stuck adjusting the
gain though.
On 4/5/06, Steve Jones [EMAIL PROTECTED] wrote:
Can you tell me what model Sangoma cards you're talking about?? The
ones I saw that had HW echo cancellation were substantially more
hardware echo cancellation).
The digium boards proved almost impossible to completely eliminate echo, and I
had random failures over time.
On 4/4/06, Steve Jones [EMAIL PROTECTED] wrote:
For phones, I've got a GS 101, a Sipura 841, and two analog phones
hooked to an GS386 ATA (one phone
From: Mike Dent [mailto:[EMAIL PROTECTED]
Sent: Mon 4/3/2006 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Frustrated with echo...
On 4/3/06, Steve Jones [EMAIL PROTECTED] wrote:
I've been using my Asterisk (At my house - 2 modem
I've been using my Asterisk (At my house - 2 modem-type
fxos, and an assortment of SIP endpoints for phones) for about 5 weeks now, and
I've been really happy with it, but I'm still having an echo problem that I've
exhausted google with, and can't get straight...
I think I've determined that
=5061
dtmfmode=rfc2833
fromuser=phone number
fromdomain=sphone.vopr.vonage.net
canreinvite=no
context=vonage_incoming
insecure=very
On 3/29/06, Steve Jones [EMAIL PROTECTED] wrote:
I know Vonage doesn't officially have a bring your own device type
program, but they do offer a softphone
I know Vonage doesnt officially have a bring
your own device type program, but they do offer a softphone. Has anyone
gotten Asterisk to connect directly to Vonage? This would be a great help!!
___
--Bandwidth and Colocation provided by
Hi, I have an AsteriskAtHome installation with
two X100p clones. Everything has been apparently fine for 5 weeks of use
or so, but today, I decided to do some tweaking of my echo cancel parameters,
and I realized that all along, one of my cards has been unavailable for
outbound calls for
Thanks for the insight.. I have been considering broadvoice because it seems
to be a big name, but yours is not the first negative feedback.. Perhaps
I'll stay w/ Vonage for the time being - I hate having it go analog and back to
digitial in the space of 5 feet of cable, but at least it works
a valid email address rather than [EMAIL PROTECTED]
Many Thanks,
Hugh
On 3/17/06, Steve Jones [EMAIL PROTECTED] wrote:
Do a tail -f /var/log/maillog which will give you a
real-time view of your mail server activity
Do a tail -f /var/log/maillog which will give you a real-time view of your
mail server activity, then while that's running, leave yourself a voicemail.
From: Tony Mountifield [mailto:[EMAIL PROTECTED]
Sent: Fri 3/17/2006 10:13 AM
To:
email has changed its policy?
Anyway, I could use a hand on how to fix this. How do I get Asterisk to use a
valid email address rather than [EMAIL PROTECTED]
Many Thanks,
Hugh
On 3/17/06, Steve Jones [EMAIL PROTECTED] wrote:
Do a tail -f /var/log/maillog which will give you a real
Although not the same driver, that's kind of the problem I'm having - It's not
that ASTERISK is limited in the AAH release, it's just that they've given you
tools to go 80% of the way without learning anything, and now I'm at a place
that I want to use some of the cool scripts I've seen on this
I have asterisk at my home, which I have typically been using about 600 minutes
outbound per month, over Vonage. I also have an old copper POTS from Verizon,
which I only use for 7 digit and 800# dialing, as well as inbound.
I am interested in setting up one or more SIP or IAX providers.
Are you selling it TO osha? If so, maybe they have an internal requirement..
If not, I've never heard of that. Granted, I haven't sold a LOT of phone
systems, but I've been involved with a couple into public works departments of
local governments as well as private corps, and nobody has ever
, Steve Jones wrote:
Are you selling it TO osha? If so, maybe they have an internal
requirement.. If not, I've never heard of that. Granted, I haven't sold a
LOT of phone systems, but I've been involved with a couple into public
works departments of local governments as well as private corps
I'm almost afraid to ask, but is the HT 386 known for having a lot of troubles?
I just installed one at home about 2 weeks ago, and knock on wood, it's only
locked up once, and this was when I was still in the process of tweaking the
config to work optimally w/ [EMAIL PROTECTED] I can't say
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Jones
Sent: terça-feira, 7 de Março de 2006 13:40
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] new beta Grandstream firmware HT488_496_386
I'm almost afraid to ask, but is the HT 386 known for having
I am starting to work on a similar solution, but with full call manager
rather than CME. I am going to use Asterisk to accept POTS calls
through PCI FXO ports (winmodems) and then forward the calls through to
call manager via SIP. I don't have my FXO cards yet (waiting for UPS
man!!) but I have
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