Northern California, bay area.
Tom Lynn wrote:
Whare are they located?
On 6/17/06, *Bob Knight* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I have 4 sparc based sun boxes I am about to pay money so I can
get rid of them. They are running older versions of Solaris.
You
Northern California, bay area.
Dovid Bender wrote:
Where are they locater ?
Dovid
*/Bob Knight [EMAIL PROTECTED]/* wrote:
I have 4 sparc based sun boxes I am about to pay money so I can
get rid of them. They are running older versions of Solaris.
You should be able to load
and mice.
Contact me offline if you want them.
I've had many good years of development on them and it kills
me to just toss them, but the office is just too damn cluttered.
thanks, bk...
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/usr/sbin/asterisk -rx 'logger reload' /dev/null 2 /dev/null
endscript
}
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work.
I'm looking for a 4-port external sip fxo which doesn't suck.
o) Clipcomm CG-410. Poor reviews.
o) Mediatrix 1204. Very poor reviews.
o) Audiocodes MP104. Poor reviews.
o) DLink DVG-3004S. Doesnt seem to exist yet.
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I am now running sip 1.6.2 with a 2.6.1 bootrom.
After moving from a 1.5 I now only see 2 softkeys at the main window:
New Call and Forward.
How do I get a Park softkey?
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the other day with a different pull
and did not have any problems. Do you think this is something related
to this box?
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Does anyone know how to silence the audible mwi on
a soundpoint ip500 or ip501 running sip 1.4.1?
I tried changing just about all the se.pat.callProg.11
vars and nothing seems to change.
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packets, looking for option names.
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me, c code. You can even use SNMP to monitor the PSTN
line status. Way cool stuff and these boxes just run forever.
If you are ready to give up on the boxes and want to dump them at
a good price, just let me know.
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I am looking for a switch that I can set up priority
queues either on a per port bases or mac address.
I really don't want to screw around with anything
above L2 or routing.
Something small (just a few ports) and cheap that I
slam in just before the dsl or cable modem.
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recommendations please?
thanks, bk...
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sipura 3k
this unit has been pretty solid, but I must admit I still do not
have it working the way I want.
There are folks that are running their own code on this unit.
It would be nice if someone would come out with an IAX version.
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let me know
I may be able to help. Contact me off list is best.
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has actually done it
You are going to have to wait for that.
http://www.eweek.com/article2/0,1759,1724923,00.asp
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for the IAX2 trunk ( and later on for Conference calls ). I
have also read about the rtc package but have not tried it.
I may have overseen very basic things... Please enlighten me!
Regards: Håkan
Try running zttest.
Once zttest is working you should be OK.
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(like 1234) it does the
normal answer thing.
If you dial an * before the extension (like *1234) it does
the auto answer thing.
So you could just use:
Channel: sip/*1234 or
Channel: sip/*pete_dest
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). Or has the
short-range DSL stuff (which I know at least one local telco uses for
in-house network extension purposes) finally beaten that speed?
... JG
15Mbps symmetrical
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gateway.
The little termination modems have ethernet and fxs.
Just add an * box, done.
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upper case only terminals, so as soon
as it sees the capital P, it will turn on folding.
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and phone calls are great.
I think it is only when I get into the pseudo zap driver that I start
having problems.
Is it time for me to check out app_conference?
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and testing when you are ready
to share.
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Steve Underwood wrote:
Arinze Izukanne wrote:
Hi Guys,
Does anyone know of E3 PCI cards that work with
Asterisk?
Arinze
No, but if you find an E3 PCI card with nice Linux support there might
be people interested in helping to get it working with *.
SBE (side band engineering).
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Jamie Carl wrote:
Bob Knight wrote:
There is a linux package called mbrowse that you can use with your
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not
spent
that much time on it.
I don't even have the MIBs which
as yet.
Thanks again guys.
There is a linux package called mbrowse that you can use with your
mediatrix mibs.
I can get and walk everything in my 1204's.
For some reason I have not had any success with writes, but I have not spent
that much time on it.
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and ready for prime time commercial deployment.
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back down to 4. It gets a little verbose.
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/var/lib/pgsql/data/postgresql.conf
change line to:
tcpip_socket = true
Add -i to option in your /etc/init.d/postgresql
export PGOPTS=-i
Not sure if all that is needed, but it did get my FC2 linux 2.6.6 running.
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, my SJLabs softphone works ok with Mediatrix any ideas?
Any working configuration?
Turn VAD off on the 1204.
* can not clock itself.
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calls.
I think the speed went down from 64 to 56 when you did this, but you
saved some per minute phone charges.
The good old days.
The phone company never seemed to really want to deal with isdn back
when it was cool. Now with dsl, they must really ignore it.
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this to compare different codecs.
The impulse function will tell you how long it takes.
The transfer function will tell you just how good a
job it did at reconstruction the original audio.
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real pretty pictures on the screen.
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that cd on a linux box.
Anyone know why?
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Stuart Baggs wrote:
Can someone please tell me what sound files to record to get wakeup.agi
to work?
I'd recommend William Hung's version of She Bangs.
If that does not wake up up, nothing will.
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-info.org in Google is an excellent resource.
Don't forget the most important link of all at the bottom of every
email. The unsubscribe link!
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off the self.
You can always use an eval board from IBM or Moto. Any expensive but
easy way to start.
The only pain would be the * port.
Yet more ifdef's. OK, that is a different rant.
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polycom reseller that will provide boot and
sip load images?
I sure like the way these phones look and feel.
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. But it would still be fun.
btw: this is not a very pretty port. The current state of the * source
tree does not lend itself
very well to other OS's. Quite a bit of hacking involved. Something
that I would never
want to see checked into cvs.
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Michael Welter wrote:
Does anyone know of a 24 port ATA device that could be installed in a
phone closet? Like a channel bank, but, instead of multiplexing onto
a T-1 circuit, it would convert to SIP/RTP on a LAN connection.
Thanks,
mediatrix 1124
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+ Batteries = Homemade WiSIP, IIRC
After a peek under the hood, I would guess we could have these manufactured
over seas for around $1000 USD per unit. It would not be the same to modify
the design in any way.
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a rough estimate so we don't wind up packed into a tiny
bar or something.
2 more bay area * nerds ready and willing to participate and help in any
way:
Bob Knight in Livermore
Todd Taylor in Tracy
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there are work arounds. I am unemployed and I do not have the time.
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how many
you want. If you wanted to get fancy, you could use a database
(perhaps with triggers?), voice recognition, or mp3s for the user to
wake up to.
Good old at job may be able to help with this (man at).
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Not sure if I will attend VON, but myself and a friend would be
way into an * nerd fest.
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Roy wrote:
Here's the web site for the convention http://www.pulver.com/von/
The convention center has conference rooms and breakout rooms. I bet if you
asked nicely, you could get one for an asterisk BOF
Yeh, but what kind of beer do they have on tap?
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or just some invalid number?
Gee, the price on those sip gateways don't seem quite so high now.
have fun, bk.
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have just been waiting for 2.03x release of the day to see if
it gets better.
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completion, calling vs called party
hold, etc, and getting rather confused watching the various packets
flowing between sip devices with a sniffer (and no reference document).
Rich
It may be a little verbose, but you can find it in the rfc 3261 as a start.
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or
something) is very handy for this.
Perfect. Thank you very much. bk...
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of info.
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really no work to do.
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:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
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linux tools I could use to do this?
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power and HEAT!
When I was making Portmasters at Livingston/Lucent we made modem boards
with a bunch of DSP's sitting on TDM's. Some of those DSP's are great
BTU generators. Some times you have to clock the DSP at slower speeds
just to
keep the heat down.
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some error handling if needed.
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that tune out of your head.
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.
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Is that FCC sticker on the back of the phone for real?
A customer could not use his computer while talking on his GS BT102 phone.
The customer was using a major name wireless keyboard/mouse with his pc.
The keyboard/mouse stops working if the GS phone is too close.
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at all.
Just unix work stations.
Of course, I could always be wrong :-)
I would not say you are wrong.
You are just looking for something different than I am.
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CW_ASN wrote:
How about to build an ip phone with this IC?
http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId=
969path=templatedata/cm/general/data/bband_ipphone_tnetv1001
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent
.
If Digium wants no part of it, we just build them on our own for our own use
or sell them on ebay.
What we would provide is schematics and source code.
Everyone can take this to their favorite fab house and crank em out.
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to large cat 3 environments and doing the
BLEC thing
bolted up to *. Very swt set up.
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end user makes it unusable.
If anyone on the list has successfully configured and used the GS
speaker phone,
could use please share
thanks, bk.
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Walker Haddock wrote:
If anyone on the list has successfully configured and used the GS
speaker phone,
could use please share
Great fix, replaced with Cisco 7960
Almost the same fix as mine, Snom 200's.
Now I just need to fix the bottom line.
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Is the party at the Paris Hilton?
sorry, couldn't help it...
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had it pointing to my * server for ntp.
I did not have ntpd running.
You could start a call, but during the call it would
do a ntp request. The server was sending an ICMP message
back and then the adapter would terminate the call.
I fired up ntpd and all is well.
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John Todd wrote:
At 11:04 AM -0800 12/5/03, Bob Knight wrote:
Greg Boehnlein wrote:
On Thu, 4 Dec 2003, Bob Knight wrote:
Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice
traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000
Greg Boehnlein wrote:
On Thu, 4 Dec 2003, Bob Knight wrote:
Steve Dolloff wrote:
I would be seriously wary of putting a DS3's worth of voice traffic on a
TNT. I don't believe they are rated to handle that much voice. The
APX1000 would be a much better platform, but I don't know if you
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not a GS bug. But SIP works well..
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Has anyone used the speakers on sip phones as part of an intercom?
Are there sip messages you can send a phone to simulate key strokes,
like someone hitting the speaker phone button on a GS?
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of letting the static box keep track off all the
dynamics and host an IAX name server.
Before I go off and so something really silly and a waste of time,
is there an easy way to do this?
Has anyone done this with DDNS?
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Sorry all you GS phone fans.
I did not read the trace correctly.
Looks like it was * that cause my double digit.
A hunting I will go..
Bob Knight wrote:
I can't afford Cisco phones, but I'll tell ya what I see with GS phones.
They seem to be bouncy as hell.
I'll hit a key and see 4 - 6 rtp
connections work fine.
I call an extension on another * box and place it on hold.
MOH over IAX/IAX2 is great.
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Juan, thank you very much.
Turning off VAD did it.
All is well.
Juan J. Sierralta P. wrote:
On Wed, 2003-11-19 at 16:10, Bob Knight wrote:
I have an * setup with sip phones and sip fxo gateway.
When a sip phone places a sip/fxo call on hold, MOH is very choppy.
It looks like RTP has a real
when a GS places
a SNOM on hold
john brown
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password protection. Hopefully they will come up with
a web browser configuration in the future.
Sean
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the syslog feature on 1204. I have the logging cranked up
to a
level 5. Now I just have to figure out what all these messages mean.
Sean P. Robertson wrote:
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 05, 2003 8:54 PM
Subject
://sitebuilder.yahoo.com -
Free, easy-to-use web site design software
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when someones commit has broke your tree.
thanks, bk
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wcfxs.o failed
/lib/modules/2.4.20-20.9/misc/wcfxs.o: insmod wcfxs failed
Any advise.
Thanks.
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[-w] the work option
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