On Tue, Jan 11, 2005 at 10:44:19PM -0300, Bartosz Jozwiak spake thusly:
We are looking for commercial solution SS7 with Asterisk.
It does not need to be build-in with Asterisk.
Could anybody suggest something?
I see a lot of people asking for asterisk and ss7. Just what exactly do
these people
On Wed, Jan 05, 2005 at 04:01:07PM -0800, Michael Swan spake thusly:
I've struggled for several days trying to get a Digium TDM04B 4-port
wxfco card working on a Dell 1U PowerEdge 750 machine running
Fedora Core 1. I finally got a call back from Digium who indicated that
snip
a power cycle
I am pitching an Asterisk solution to a local company when a few
interesting questions came up.
What is the max # of callers that can participate in a conference call? A
meetme style call, if it makes any difference. I have googled and looked
around and cannot find any limit.
Also, is there any
On Sun, Mar 07, 2004 at 01:20:36PM -0600, C. Johnson spake thusly:
I have anywhere from 15, to a peak max of 30
traders all using a meetme conf during the day. My
Thanks for the info. I'm really look for the hard maximum. I doubt cpu
issues will be a problem. I am just wondering if there is a
I notice that CONF_SIZE in asterisk/apps/app_meetme.c is set to 160.
CONF_SIZE is used to set up a buffer of some sort. Can anyone explain to
me why the value of 160 was chosen? Seems kinda arbitrary.
--
Tracy Reed The attachment is a digital signature.
On Wed, Jun 02, 2004 at 02:01:49PM +1000, Shaun Ewing spake thusly:
exten = xx,1,Dial(IAX2/[EMAIL PROTECTED]/phoneSIP/phone|60|r)
By that example, you can see that I am dialing IAX2/[EMAIL PROTECTED]/phone and
SIP/phone at the same time with ring back with a timeout of 60 seconds.
Note
On Sun, Nov 14, 2004 at 09:50:28AM -0700, Paul Fielding spake thusly:
Whatever. I find it frankly more annoying to have people bottom post. I
use Outlook Express for my mail (as do millions of others), and the way OE
formats it's mail lends itself to top posting.When you bottom post, I
On Wed, Nov 17, 2004 at 10:27:38AM +1100, Duane spake thusly:
Shekhar Prasad wrote:
an IAX adaptor. Everything seems to work fine. However, I would like
the dial tone signal to be generated from Asterisk when Budgetone is
picked up. It generates its own signal and does not really obtain it
On Wed, Nov 17, 2004 at 01:13:57PM -0500, Noah Miller spake thusly:
On our current phones (Iwatsu) we have a button on the
phones for each extension that lights up when that
This seems to be a popular request these days. Most places I've seen
call this shared lines I thought this was
On Wed, Nov 17, 2004 at 05:07:33PM -0500, Bob Willock spake thusly:
I just bought a couple of these Cisco 7970G phones and it seems that they
require a SIP image binary file to load when the phone boots and this file
Cisco phones are hugely overrated. I have deployed a number of them and I
have
I spent today trying to get openh323 working with Asterisk 1.0 on my new
AMD64 box. I ran into a number of problems. The first being that the
openh323 build scripts do not recognize x86_64 as an architecture that it
builds on. I hacked the scripts appropriately and got it built. I set the
On Thu, Nov 18, 2004 at 04:30:24PM -0600, Michael Shuler spake thusly:
the VoIP devices and to the media gateways. The SER machines don't know
what to do with a call they only know to hand it over to Asterisk for
routing/CLASS features or whatever you want the call to do. You then have a
I have an asterisk box with a public IP for people on the Internet to
connect to. I also have a Lucent TNT on the same physical network but on a
10.0.0.0 subnet. It isn't safe to put the TNT on a public IP address and I
never want it to talk to the net directly anyhow so this seemed like a
good
On Fri, Nov 19, 2004 at 07:44:34AM -0600, Tim Jackson spake thusly:
canreinvite=no ?
I already thought of that and canreinvite is already set to no. I also
know about bindaddr and localnet but neither of those do what I want
either. Thanks.
--
Tracy Reedhttp://copilotcom.com
This message
On Thu, Nov 04, 2004 at 04:20:53PM -0600, Matthew Boehm spake thusly:
But yes, RealTime is very nice. It is still in development and there is
progress to bring it to as many apps as possible. You can get the RealTime
MySQL driver here:
app_realtime seems a lot like res_data which is also
On Sat, Nov 20, 2004 at 06:55:56PM -0500, Noah Miller spake thusly:
This does seem to be a common request, but I haven't seen any great
Yes, it is. I am surprised * still can't do it.
Another option is the Flash Operator Panel, you can see a live demo at
http://www.asternic.com/ It is a
On Sat, Nov 20, 2004 at 09:25:38PM -0600, Brian Roy spake thusly:
I would look at putting a dual monitor on her desk. You can pick up a
15 flat panel and a video card for about the same cost as the SNOM.
She doesn't want another monitor.
Asterisk is not your dad's pbx.
Most customers don't
On Sun, Nov 21, 2004 at 12:05:27AM -0500, Gregory Junker spake thusly:
What is the size of the current line panel on her desk? I am thinking it
might be worthwhile to produce an addon to Asterisk that drives a flat
touchpanel that does the same thing as the current solution. Baby steps.
I
On Sun, Nov 21, 2004 at 06:18:04PM +1300, Matt Riddell spake thusly:
Me and another guy are working on LCD drivers etc for Linux. The thing
is, the display would be run from your Asterisk Server. I.E. It will
need to be run from either Parallel, Serial or USB port. We will open
What would
On Sun, Nov 21, 2004 at 12:21:16AM -0700, Kevin P. Fleming spake thusly:
And there are tons of extremely small systems that could do this job. I
have here in front of me a Soekris net4801 which is tiny, noiseless
I know there are plenty of small systems that would be great. The problem
is the
On Sun, Nov 21, 2004 at 04:25:39PM -0800, Tony Vickers spake thusly:
What VOIP Phones is everyone using and why? Is the a common phone that
seems the work the best? Just wondering.
My picks in increasing order of quality and price:
Grandstream - It's cheap and looks funny but it works.
Snom -
On Mon, Nov 22, 2004 at 04:17:15AM +0200, Philip Trauring spake thusly:
Can some people post some configurations they've implemented when
deploying an * system for let's say 25-50 stations and maybe a larger
200 station system? I would assume some kind of chassis with some DSP
For a 25-50
Has anyone here done this? I got it compiled just fine but when I make a
call I do not get any audio going either way. The * box is not behind any
sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I
have it set up properly to work through NAT and it will talk correctly
with
On Mon, Nov 22, 2004 at 10:00:48AM -0500, Paul Rodan spake thusly:
I am quite interested in this as well. I didn't realize registrations are
the #1 cause of load on an asterisk server, we haven't gotten to that kind
of usage just yet.
I don't think they are, are they? How could a few
On Tue, Nov 23, 2004 at 12:42:07PM +0100, administrator tootai spake thusly:
with my other regular x86 box running H323. One odd thing I note is that
when looking at the UDP traffic with tcpdump I see the * box receiving my
Same problem here. My * box is connected to GnuGK. CVS Head 11/02/04,
Ok, I have some more info. The code from openh323.org will not compile on
x86_64 but the latest from the OpenH323 project on sourceforge will
compile just fine on x86_64. Asterisk 1.0 will not compile with this new
openh323 code but it looks like the latest cvs-head does. In
channels/chan_h323 I
We are in urgent need of some help getting Asterisk to gateway between an
incoming H323 connection and SIP to a Lucent TNT. We have the incoming
H323 already set up and the SIP going to the TNT but the media stream is
getting lost somewhere as no audio is heard. We are willing to pay $$$ for
an
On Mon, Nov 29, 2004 at 04:07:36PM -0600, Steven Critchfield spake thusly:
Better question is why do you feel there needs to be a change?
Probably because he has not figured out mail filtering.
--
Tracy Reedhttp://copilotcom.com
This message is cryptographically signed for your
Some of you may recall that I have been working on building a box to
convert H323 to SIP. After a significant amount of outside help and
slicing and dicing of the ohh323 code to get it to compile on AMD64 we
finally got it working. Now we are working on improving the performance.
This box takes
On Tue, Nov 30, 2004 at 10:34:00AM +0100, Jan Baggen spake thusly:
When I have 2 active SIP calls on the 7960 phone there
are no available lines for additional calls. I tried
to configure 2 lines to the same SIP server but it's
still limited to 2 calls. How to utilize all lines?
Are you using
I am so far unable to get the busy lamps on a Snom 220 to work either with
current cvs or asterisk 1.0.
I am using the hint extension and the Snom 220 just as described in the
mini-howto on:
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html
There are also a couple of
On Sat, Dec 04, 2004 at 11:51:24AM +0100, Peter Svensson spake thusly:
I guess it may just be a typo during retyping, but you have 'l' (lower
case L) in the hint line and a '1' (one) in the macro line.
SON OF A [EMAIL PROTECTED]@[EMAIL PROTECTED]@#^*$#%@@
ahem
You are correct, somehow when I
On Sat, Dec 04, 2004 at 08:03:03PM -0400, Cian O'Sullivan spake thusly:
She is an older lady and does not want to use a web interface. Any
suggestions?
Give her a Snom or Polycom phone which does have this capability and set
it up like this:
On Sun, Dec 05, 2004 at 12:13:07PM +0800, Ronald Wiplinger spake thusly:
I want to play around with post billing. List of all phone calls, ...
Which program is useful for that?
All what I have seen are not based on CDR, but on Radius.
What are you using?
Everyone pretty much writes their
On Mon, Dec 13, 2004 at 12:50:54PM -0600, Gerald J. Puhl spake thusly:
I have been looking to see if this type of phone can be implimented in
*. I have found nothing conclusive. Is any out there using a multiline
/ mutlifunction phone typically used by a receptionist for transfering /
Has anyone been able to make the multitech voip box speak H323 with
asterisk? I am using the asterisk CVS from a week ago and the recommended
versions of pwlib and openh323. I am able to connect to the multitech 800
box at our remote office which is connected via POTS to a proprietary PBX
system.
On Tue, Dec 14, 2004 at 08:40:10PM -0500, Info spake thusly:
I had a problem with transfers on my Snom 190 until I made sure that the
Break Key setting on the advanced page of the Snom web configuration was
set to off. Then blind transfers started working great.
Well, that fixes part of the
On Wed, Dec 15, 2004 at 08:11:43PM -0600, Brian K. Hershey spake thusly:
I don't think it's the snom, (the break key is set to off)
the # key is not being interpereted by the PBX as an attempt to
initiate a transfer.
Is this an error in my extensions.conf?
Note that my Snom 220 does not have
On Thu, Dec 16, 2004 at 11:35:22AM +0530, Ashish Shinde spake thusly:
How can I solve this problem of voice quality? Can a better
implementation of jitterbuffer with packet loss concealment help? If
so how do I get the newer implementation. I would really like to help
out in the
On Tue, Dec 14, 2004 at 07:44:51PM +0100, Bruno Hertz spake thusly:
I'm considering that board as a mail and voip gateway for home use.
In view of all those statements about how little resources asterisk
needs, did anybody already try running asterisk on it?
I know these things have geek chic
On Tue, Dec 14, 2004 at 11:21:06AM -0500, Goutam Shaw spake thusly:
I upgraded to 1.0 release recently from 0.9 and noticed that * is running at
99% CPU. Is there a known problem?
Does it say anything about the console sound device not working or
something when asterisk starts? I once had a
On Tue, Dec 14, 2004 at 06:14:59PM -0500, Andrew Kohlsmith spake thusly:
On December 14, 2004 04:11 pm, Greg - Cirelle Enterprises wrote:
It's been hours since I've seen a post from this list
Must be broken again.
So you'll email a broken list to send a message...? :-)
Sure, why not? You
On Thu, Dec 16, 2004 at 01:11:34PM +0100, Roy Sigurd Karlsbakk spake thusly:
I think there is a bit more difference. The byte code of ulaw is a
monotonic function of the amplitude whereas in alaw the code is xor:ed
with a bit mask of 0x55.
Wow! Encryption!
Scary thing is, it would be
On Sun, Dec 19, 2004 at 12:09:48PM +0200, Shoval Tomer spake thusly:
Is there a way to use asterisk for call screening?
Meaning, a call comes in, asterisk answers with voicemail after I don't
pickup, and the voicemail prompt + the caller's message a played via the
sound card on asterisk. If
I am having a hell of a time with transfers.
First the Snom issues:
The transfer button on the Snom 220 does not work. I have read about
setting break key off in the advanced page of the web config but the Snom
220 has no such option. At the moment I am having to use the # transfer
hack which
On Tue, Dec 21, 2004 at 11:05:27PM -0500, Alex Brecher spake thusly:
I still don't get why we don't move over to a web based forum ? I can set
one up on a dual athlon server with 4Gb of memory if you guys are interested
Because it is MUCH nicer to be able to read through articles after they
are
On Wed, Dec 22, 2004 at 09:58:53AM -0700, Kevin P. Fleming spake thusly:
There are SIP transfer problems in CVS HEAD at the moment, although a
fix is on the way (monitor bug 3113 in Mantis).
Thanks. Last night I downgraded to CVS-v1-0-12/21/04-16:31:58 just in case
this was the issue but I
On Wed, Dec 22, 2004 at 09:58:53AM -0700, Kevin P. Fleming spake thusly:
There are SIP transfer problems in CVS HEAD at the moment, although a
fix is on the way (monitor bug 3113 in Mantis).
Not only in HEAD but also in STABLE, oej just informed me. I thought I was
losing my mind when I went
On Wed, Dec 22, 2004 at 01:33:35PM -0500, Alexander Lopez spake thusly:
OK with CVS-HEAD-12/22/04-09:47:32 incoming (to ata) sip calls on a
ATA0186 now seam to work fine. However transfers still do not work.
With CVS-HEAD-12/22/04-12:46:47 transfers still do not work.
Turns out I
On Mon, Apr 12, 2004 at 02:19:23PM -0400, James H. Cloos Jr. spake thusly:
Yes, there would be. This is the same issue as using nfs mail spools
with maildir style storage. W/o locking there is no way to guarantee
that two servers do not create the same vm file on top of one another.
The
On Fri, Apr 16, 2004 at 06:04:04PM +0100, Craig Waddington spake thusly:
When we receive or make a call to the outside - they can hear us, but we
cant hear them.
I have had this problem several times and so far no resolution. However
for me it has always been with IAX. I have been told that IAX
On Sun, Apr 18, 2004 at 09:31:48AM +1000, Duane spake thusly:
be sure more are issued on a correct basis. PGP model if you lived in
say Africa and wanted to communicate with someone in South America with
little or no prior relationship and you wanted to be sure the
communication wouldn't be
On Sun, Apr 18, 2004 at 10:22:08AM +1000, Duane spake thusly:
Just a little matter of key distribution, how do you know the CA key
given to you is actually the CA? Especially since Thawte no longer does
PGP key signing and verisign is making too much money from PKI...
Same way I know
On Sun, Apr 18, 2004 at 11:13:27AM +1000, Duane spake thusly:
But have you ever met face to face with an employee from a CA and
verified they were an employee or just grabbed the info from their
website and assumed there was no man in the middle attack sending you an
alternate
On Wed, Apr 21, 2004 at 09:20:54AM +0200, Altus Snyman spake thusly:
I'm still looking for a SIP client that will work on fedora core 1?
Thanks
linphone? www.linphone.org
--
Tracy Reed The attachment is a digital signature.
http://copilotconsulting.com More info:
On Thu, May 13, 2004 at 10:57:30AM -0400, Joseph spake thusly:
Is there any way to play background music on a sip phone
while the phone is not in use like many legacy pbx's offer?
Why would you want to? The sound quality is horrible for music even on a
good speakerphone. You've probably got a
On Sun, May 02, 2004 at 12:37:14AM +0100, Gavin Hamill spake thusly:
On Sunday 02 May 2004 00:32, you wrote:
How about setting up a bounty?
http://voip-info.org/wiki-Asterisk+bounty
If I had the money to rent-a-coder, would I have begged on a public mailing
list?
You are missing the
On Tue, May 25, 2004 at 05:53:55PM -0500, Roger spake thusly:
Thanks for the reply - I have version cell phone service. I did a work
around and called my cell phone via IAX2 as opposed to the zaptel
channels. This works and all 3 extensions ring w/ no problem.
I am having the exact same
On Fri, Nov 12, 2004 at 06:57:05PM -, Kevin Walsh spake thusly:
up properly. There is no excuse at all for lazily top-posting.
As a businessman I also see it as a matter of professionalism. I see top
posting and not trimming etc as just unprofessional. I regularly do get
poorly formatted
If anyone is interested in some used digium hardware for their projects:
T-1 card:
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4 port FXO cards:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5102115738
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5102264062
Tracy R Reed wrote:
If anyone is interested in some used digium hardware for their projects:
T-1 card:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5101873998
4 port FXO cards:
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=5102115738
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem
For the last couple of days I have been battling a * installation (company
PBX) that has been spinning on the cpu at 100% utilization. This was
causing dropped calls, horrible SIP call quality, etc. The box is running
the CVS * as of Aug 5 on Fedora Core 1 on an AMD Duron processor. I called
When I try to flash my 7960 with SIP I get messages like this in the tftp
server logfile:
Aug 27 02:01:17 home tftpd[32590]: tftpd: trying to get file: P0S3-03-0-00 .bin
and the phone says something similar on the display for a brief moment and
puts a funny char where the space in the filename
Anyone using Python to write their AGI applications? I find very little
info on it. The wiki has a link to http://sourceforge.net/projects/pyst
but it seems like a dead project. I posted some questions to their mailing
list a week ago and have not seen a reply or other posting. Is there some
other
On Mon, Sep 06, 2004 at 10:40:44AM +0300, Elman Efendiyev spake thusly:
I believe it's should support T.38 in pass-thru mode. I mean setup
like this:
I have seriously been considering giving this a try lately also. I will be
very interested to see how it works out. What sort of problems exactly
On Mon, Sep 06, 2004 at 01:34:50PM +0300, Elman Efendiyev spake thusly:
Jusc couldn'n transmit faxes trouth asterisk. It just hangs up when
starting a fax transfer
If U will do some experiments with it I would be happy to hear any
reslts/info
I have given this some more thought. Fax Protocol
=1item=8738074150
--
Tracy R Reed
http://copilotconsulting.com
1-877-MY-COPILOT
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