On Dec 31, 2005, at 7:28 AM, Ross C wrote:
Peter,
After upgrading to 1.0.1.13 I had some miscellaneous problems on
one of my
GXP-2000's--it would grab an IP address, but it wouldn't get the
time/date,
it wouldn't register, blah blah blah. I could access the web
interface OK,
so it
Kristof Hardy wrote:
Was there a resolution to this issue? The GXP-2000 seems to be a very
popular phone, so I can't imagine others on the list not experiencing
this? Or is this part of a batch with unresolvable problems that I
need to send back to the seller?
Well, I'm using dozens of
Tony Hoyle wrote:
Philip Edelbrock wrote:
18 17.161118 Grandstr_05:a9:bf - BroadcastARP Who has
206.228.191.144? Gratuitous ARP
19 17.609869 3com_96:2f:eb - Grandstr_05:a9:bf ARP 206.228.191.144
is at 00:10:4b:96:2f:eb
20 20.155260 206.228.191.144 - 206.228.191.7 DHCP DHCP
iaxmodem + hylafax worked much better for me. Seems solid where
txfax/rxfax was very iffy. Thus far, I'm just using it with Zap lines,
though.
Phil
Technical Support wrote:
Downgrade your spandsp. Do some reading on spandsp first!
-Original Message-
From: [EMAIL PROTECTED]
Jeff Herring wrote:
would you care to share with the list your installation procedure
and configuration files associated with your iaxmodem and hylafax
installation alongside asterisk?
Sure! Some things, I'm sure, could use improvement, but this is working
for me:
Get iaxmodem:
We're having a problem with call screening with our existing legacy
system (Toshiba DK40i) which the touch-tone buttons don't work when *
calls extensions. At first I had set up a 'press 1 to accept' prompt,
but it won't work if the DTMF buttons aren't functioning, of course.
So, a
I've got lines coming in from a legacy system (into FXO ports) which
does not give any disconnect notification. Folks familiar with the
system say that I can buy or build a device which will listen for so
many seconds of dead air and then automaticly send a disconnect signal
to free up any
We're got a T1 from Sprint that we use for internet. During VIOP calls,
if you download something, the VOIP calls break up.
I found some info at Sprint for adding 'class of service', and I also
have some information on configuring our Cisco routers.
I've read the relevent pages on the
Rick Smith wrote:
Phil;
What link ?
Your question is a bit vauge, but here are some relevent urls:
Sprint CoS request form (a 2 pager, with some great links to a
guidelines doc and faq):
http://www.sprintlink.net/maint/cos_template.cgi
QoS:
http://www.voip-info.org/wiki/view/QoS
Phil
David Choo wrote:
Hi,
Consider doing rate limiting / bandwidth reservation. It worked heaps of
wonders for mine!
That's good to hear. BTW- Am I doing this right? Here are the relevent
chunks of my config on my router:
!
!
class-map Platinum
match access-group 101
!
!
policy-map
IPCOS
But, on their end (as an output). I thought about adding an 'input' on
my side, but it seems like it's too late at that point since it's
already traveled through the bottleneck (T1).
Ideas?
Phil
Philip Edelbrock wrote:
David Choo wrote:
Hi,
Consider doing rate limiting
Michael Collins wrote:
Just curious to know if anyone uses Festival with * and whether or not
you’ve got a different voice than the default. I’m looking at doing a
commercial application but my boss doesn’t want to shell out the $
before we do some real world testing of * and Festival.
Clint Sharp wrote:
I'm still having numerous echo issues, even on SIP calls, with the
GXP-2000s. Unfortunately, they cause echo on the remote end on SIP
calls, which does not occur on other phone models. The speaker phone is
unusable due to echo problems. Maybe the 1.0.2 firmware branches
On Feb 18, 2006, at 11:35 AM, J Poz wrote:
I have a specific business problem that I'm hoping someone has
ideas and/or has already worked out a solution.
My application needs to be able to automatically create and issue
faxes to many different fax machines. The volume is going to be
Marc Archer wrote:
Can someone give me a definite answer as to wether or not you can
reliably run multiple TDM400P’s in the same machine?
I need 4 x FXO and 4 x FXS to connect to both the PSTN and existing key
system, but I have seen several threads suggesting that this is not a
supported
Rich Adamson wrote:
I want to sniff all these info to test a sip ip phone talking to a asterisk
server. I have used tcpdump, but It just shows the
Ethereal would probably be a batter analyzer. Not sure how well it
seppurts sip, though. Unlike tcpdump it won't work on-the-fly. But you
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it
charge before I can play with it.
A few quick comments:
- I started a Wiki page at voip-info to post issues, firmware news, etc.
I really like the wealth of info on the GXP-2000 page, so I wanted to
start something
Philip Edelbrock wrote:
Whoo hoo! I just received my WIP300 from voipsupply. I have to let it
charge before I can play with it.
After it charged and I started using it, I had three crashes. Once
during a call (exactly 3 minutes into it, according to the frozen
display), and twice
Omar A. Sabek wrote:
Like BJ, I'm sorry you had bad luck Phil. I have been playing with
this phone all weekend, and I have had minor problems. The voice
quality is as good as my cisco and polycom sip phones. I asked a
friend to guess what kind of phone I was talking on and he said it
sounded
Charles Marcus wrote:
[...]
So, how much work are we talking about to get our current system to play
nice with Asterisk? Will we lose any functionality? Gain any? Do you
know of any technical how-to's that my phone guy would be able to answer
these questions from? Are you available to
Something I've been curious about is if it is possible to stick their
ata on a extra ethernet port on an Asterisk server and have the Asterisk
server spoof the Vonage server. Then, do a man-in-the-middle type thing
to use the ata for authentication, but have Asterisk handle all the calls.
I experienced this today. Doing a 'show channels' in Asterisk showed a
Zap line perpetually ringing the sip phone even though the sip phone was
reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip
allowed 2-way audio to resume.
Phil
Frederic Jean wrote:
Hi Geoff,
You
I've got a voicemail server I made from four X100P cards (off eBay),
Fedora Core 4, connected to a Toshiba DK40 system. I'm using
Asterisk 1.0.9, and Zaptel 1.0.9.2.
It works great, except the card which receives a majority of the
activity occationally will go into a 'Red' alarm and
On Nov 13, 2005, at 4:31 PM, Noah Swint wrote:
Are you running off the rpms or compiled version?
Compiled. Actually, I had to compile and install it twice because
the first time I didn't have Zaptel installed (which needs to be
installed first, apparently).
Do you suppose it makes a
On Nov 13, 2005, at 6:09 PM, Rich Adamson wrote:
About a year and a half ago when I was running a couple of x100p's
there was an issue associated with disconnecting the pstn line from
the card. If I recall correctly, if the pstn line was removed for
more then a second or so (a couple of
Tzafrir Cohen wrote:
[...]wcfxo is the driver for the X100P cards.
FYI-I just had another crash. This time I got an oops dump:
[ cut here ]
kernel BUG at mm/rmap.c:493!
invalid operand: [#1]
Modules linked in: loop wcfxo(U) zaptel(U) crc_ccitt ipv6 parport_pc
Logan wrote:
I was wondering if it was feasable to istall
Asterisk on this box and have that modem (or whatever modem) with a
regular telephone wired to the Phone port.
I'm a bit of a noob, also, but I don't think the Phone port on those
cards are real FXS ports. I.e., I think they just
Logan wrote:
Hi everyone!
Okay. I was reading on the voip-info.org about FXO and FXS. Is it
possible just to get a card with FXO and FXS together? I know Digium
sells them, but as I've said, I'm looking to spend too much.
Thanks for everyone's input!
Logan.
FXO is easy, but FXS is more
I'm curious if anything new has been determined on this phone? Is it
SIP compatible with Asterisk and, say, Broadvoice?
I'm a little wary that this may be vaporware. The phone doesn't seem to
be listed by the FCC. But, I would preorder one if it's Asterisk and
Broadvoice compatibile.
Bruce Ferrell wrote:
Hi all,
I just aquired some new SIP phones as gifts from a friend, a Uniden
UIP200 and UTstarCom Wifi F1000.
Unfortunately neither came with information about how to configure them
remotely.
From what I see on the UTstarCom user forums if the phone comes from
Sorry, this is slightly off topic, but I wonder if somebody has some
hints on getting our Meridian system to output DTMF tones to our
Asterisk box. Simply put, when buttons as pressed, nothing happens.
The Asterisk box has a 4 port Digium FXO card.
This is what we've got:
Meridian
On Dec 15, 2005, at 6:23 PM, Steve Totaro wrote:
Sorry, this is slightly off topic, but I wonder if somebody has some
hints on getting our Meridian system to output DTMF tones to our
Asterisk box. Simply put, when buttons as pressed, nothing happens.
The Asterisk box has a 4 port Digium
On Dec 18, 2005, at 12:01 PM, Andrew Kohlsmith wrote:
On Sunday 18 December 2005 14:32, Mohammad Shokuie wrote:
As a matter of fact im serious to know where is the source of echo
in a
pure VoIP connection, i think the most of echo problems come from
hybrid
circuits which are not an issue
We're using a Budgetone 101 ($60) SIP phone. It works pretty well. No
echo cancellation, though, which is a little annoying when used
somewhere with significant ping-times to the server.
Phil
Rehan Ahmed wrote:
Hello Dakota,
I have a few that i can ship you from vida21.com
This is more of a curiosity and a thought than serious issue. But, I
wonder if I can get my Asterisk server to authenticate to my provider by
throwing the authentication requests to the SIP analog-adapter they
shipped me? (And I can't get in and see the authentication credentials
in the
Ron Bulthuis wrote:
I just purchased a Grandstream gxp-2000, budgetone102 and a HT-386.
Browsing to each device by IP address, I can get logged in using admin
and I can see the advanced settings, however, if I try to change the
settings and clicking the Change button, it just brings me back
We've done a direct swap of an old Amanda voicemail system with a shiney
new Asterisk system (Asterisk 1.0.9). The system consists of 4 FXO
ports on the * box (TDM400P), and three old Wildcards we aren't using
(too buggy we found).
CO lines- Toshiba - FXO ports on *
We want to branch out
Michael Sampson wrote:
I'm not really trying to monitor anything on the asterisk box at all. I
guess this is more of an SIP phone question. Really all I need is to get
the audio from an SIP phone, both the caller and callie, to a 1/8th inch
stereo jack that I can plug into a mic input.
We're getting our feet more and more wet with VOIP at work. We want to
experiment with a good wireless (as in WiFi) phone. What would be a
good phone to impress my boss with?
I'm personally drooling over the UTStarcom F3000, but compatibility and
shipping ETA info is a bit sketchy.
Works fine, good batt. Live.
Decent sound quality.
All in all a good product for about 150 euro's
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Philip
Edelbrock
Sent: Tuesday, January 10, 2006 2:45 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk
We've got a Toshiba DK system w/ analog ports that went to a
voicemail server. I swapped in an Asterisk box with a Digium 4-port
fxo card. It /almost/ worked perfectly.
The problem is that Zap channels never hang up. They have to time out.
I set up MeetMe, but all Zap channels hung
Jim Freeze wrote:
[...]
So for 5 phones, I would need 2 cards. And, the O'Reilly book says that
I should not put 2 cards in the same box, so I would need another
computer.
[...]
Whoa, I'm confused. Can't you use as many cards as you have slots?
We've got just one 4-port card, but I've
Darrick Hartman wrote:
A little background. I'm integrating asterisk as the voicemail service
for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2
is used to connect an analog device (such as a voice modem) to the pbx.
In the past we've used vgetty and a voice modem with
Ben Fried wrote:
On 1/9/06, Rich Adamson [EMAIL PROTECTED] wrote:
Sorry in advance if this is a FAQ...
I've got a working Asterisk setup based on [EMAIL PROTECTED] 2.2. I have a
TDM400
card with 2 FXS and 2 FXO ports; PSTN connections come in via the TDM
card.
I haven't been able to get
Ben Fried wrote:
Since writing my message, I appear to have had success using iaxmodem
+ hylafax to do inbound faxing. Setup was not completely obvious,
especially if you're usin [EMAIL PROTECTED], like me, I finally seem to have
inbound
faxes working properly now - 5 or 6 in a row have all
Ben Fried wrote:
Just to be clear, I have inbound and outbound faxes working with my
TDM400, by going the iaxmodem and hylafax route. No need for a
separate modem or an x100p card.
Be
Ah, that's interesting. Can you provide some details on how you set it up?
Phil
Random thought: They look like they are owned by asterisk. Are you
running zttest under an asterisk account or as root?
Phil
Antonio Moragues wrote:
The device is in place:
# ls -l /dev/zap/
total 0
crw-rw 1 asterisk asterisk 196, 1 Jan 19 23:41 1
crw-rw 1 asterisk asterisk
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