Jon Pounder wrote:
Who said I wanted to run DSL over it :)
no one - I'm sure you really just want to run 110baud modem over it :)
and I'm sure you probably don't want a handful of them between the same 2
locations either.
btw - here is an interesting strategy to get fibre or something
Alex Balashov wrote:
On Fri, 11 May 2007, John Treble said something to this effect:
Can you still do “homebrew” PTP T1 in the U.S. this way? I thought
this was nixed by the ILEC/CLECs years ago.
It's logically possible. But if you're trying to do T1 over a single
pair, you'd have to
Andrew Kohlsmith wrote:
On Friday 11 May 2007 5:45 pm, Jon Pounder wrote:
again, I'm interested to know anyone whose actually done this, and what
the results were, since I have been thinking of the same thing for a
while.
I'd run about two dozen of these things using a variety of equipment.
On Sat, 12 May 2007, Stephen Bosch said something to this effect:
The copper is so decrepit in so many places that it's really
questionable whether this is even worth considering.
There are just too many variables outside your control.
That was my general impression as well. My suggestion
Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but
From: Mike [EMAIL PROTECTED]
Date: Fri, 11 May 2007 19:44:51 -0400
Yeah ok. That doesn't help.
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
ChanIsAvail() is supposed to allow this.
Yuan Liu
I want it to ring 30 seconds and then Hangup if
I am curious if it is advisable to use implement Asterisk as a Session
Border Controller for a VoIP reseller environment. Users will terminate
calls SIP to my server, which will authenticate them via RADIUS, perform a
LCR lookup, select an appropriate trunk (based on LCR), and terminate the
call
Greetings,
It is my impression that Asterisk cannot safely handle more than about
100-200 calls in parallel, but it may be possible to increase the yield by
removing any transcoding and offloading some of the channel functionality
to hardware DSP boards. I do not know much about this, and
On Fri, May 11, 2007 at 11:08:50PM -0400, Jonathan Addleman wrote:
pedro noticioso wrote:
hi there guys!
how can I eliminate this message?
[May 11 11:00:46] WARNING[7039]: res_musiconhold.c:506
monmp3thread: Unable to spawn mp3player
[May 11 11:09:06] WARNING[7039]:
Dear
I am using ser + asterisk, for setting up land line
calling.
only probelm, each unregistered soft phone can places
the call only with callerid,
this is critical problem, because any number(soft
phone) , has a limit time to use this system,
best
Mani
Hi,
My SIP phones which are working fine at all other remote locations, when
placed in a location connected to the Internet through Rogers Wireless
Internet, they don't work. They do get registered on low port numbers, like
, 1123, 1231 etc. but can't dial out or receive calls. Asterisk CLI
Yea that was my first guess until I saw the packet dump prove out
that the ATA was transmitting it. Eh, let me go searching through
the bug lists see if I can find something in older versions.
On May 11, 2007, at 3:04 PM, Matt wrote:
I have actually seen this behaviour on 1.2.x. I always
hi,
i am updated to latest asterisk stable (because of security problems), but
now asterisk crashes within a hour
log is clear
do you someone have this problem too?
---
Marek Cervenka
===
Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't real PTP copper and
therefore anything but
hi all,
i have problem with dtmf detection on wctdm24xxp with full fxo and vpm module.
after pushing dtmf tones on my phone for several times the card just
detects one or two digits randomly.so now i can't use any voice menu
on my box with this card.
i have tried the following scenarios:
- the
Jon Pounder wrote:
that's what dry copper is supposed to be, just a cross connect between 2
pairs out of the CO. ie not even battery, line test equipment, or anything
else hanging off it at the CO. any restriction should be purely a function
of the inductance/capacitance of the wire and the
Greetings list,
I've been having a go at preparing some music on hold from CDs clients have
supplied, but quality seems really rather poor over compressed channels (tried
g729, GSM and Speex). I've been doing the following:
sox -v 0.15 filename.wav -t raw -r 8000 -s -w -c 1 filename.sln
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't
Tzafrir Cohen wrote:
It actually is (maintained, and a recent version of it is in
stable/testing.
Hmm.. I think several years ago it wasn't... I guess I'm just living in
the past. Sorry about that!
--
Jon-o Addleman - http://www.redowl.ca
___
Quoting Eric \ManxPower\ Wieling [EMAIL PROTECTED]:
Jon Pounder wrote:
that's what dry copper is supposed to be, just a cross connect between 2
pairs out of the CO. ie not even battery, line test equipment, or anything
else hanging off it at the CO. any restriction should be purely a function
Quoting Stephen Bosch [EMAIL PROTECTED]:
Jon Pounder wrote:
On 5/11/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Fri, 11 May 2007, C F said something to this effect:
Not according to Verizon (in my area anyhow), We tried it and it
didn't
work. The verizon technician insisted it wasn't
Eric ManxPower Wieling wrote:
Jon Pounder wrote:
that's what dry copper is supposed to be, just a cross connect
between 2
pairs out of the CO. ie not even battery, line test equipment, or
anything
else hanging off it at the CO. any restriction should be purely a
function
of the
Hi!
Is it just me or do the last 2 or 3 versions of the zaptel-1.2 branch seem
to break cli? Often not the full number is displayed, or only 2 or 3
digits?
I am in The Netherlands, and have had this in my zapata.conf (which used
to work flawlessly) :
signalling=fxs_ks
immediate=yes
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted it.
And it's quite a hoot
Thanks Alex, for the quick and detailed response. I really appreciate it.
Well, that's pretty much what I've been getting from my reading.
I'm curious as to what areas (specifically, modules/features) of OpenSER I
should look into to accomplish this.
Essentially I don't want any endpoint to
On Sat, 12 May 2007, Atlanticnynex said something to this effect:
'SIP Redirect Proxy', which I'm understanding to just redirect the SIP
requests to the appropriate destination based on the routing logic- and
OpenSER is no longer involved in the call process (meaning that
accounting can no
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted
it.
And it's quite a
Thanks Alex, some great ideas.
I think, however, I'm leaning towards Asterisk at this point- since I have
quite a bit of experience there, and very little with SER. At this point,
I'm wondering from a dimensioning standpoint, what kind of capacity my
machine will have (Dual Core Xeon 2.4GHz 4GB
I've reposted to a new thread Asterisk High Capacity Stability as I don't
think the subject of this thread is appropriate anymore.
-kn0x
On 5/12/07, Alex Balashov [EMAIL PROTECTED] wrote:
On Sat, 12 May 2007, Atlanticnynex said something to this effect:
'SIP Redirect Proxy', which I'm
Hi,
I am currently using TE110P Digium card on a PRI card. Basically the
echo is so much that one can disticntly identify that. I have tried all the
combination if tuning configuration seen in forums etc. I am using MG2
cancellor algorithm also tuned the RX TX gains, still there
On Sat, 12 May 2007, Atlanticnynex said something to this effect:
Thanks Alex, some great ideas. I think, however, I'm leaning towards
Asterisk at this point- since I have quite a bit of experience there, and
very little with SER. At this point, I'm wondering from a dimensioning
standpoint,
Deepak Naidu wrote:
Hi,
I am currently using TE110P Digium card on a PRI card. Basically
the echo is so much that one can disticntly identify that. I have tried
all the combination if tuning configuration seen in forums etc. I am
using MG2 cancellor algorithm also tuned the RX TX
On Fri, May 11, 2007 at 11:04:12AM -0300, Juliano Fernandes Schroeder wrote:
I'm trying to configure a TDM808P card on debian. When I modprobe wctdm24xxp
i get this error
FATAL: Module wctdm24xxp not found.
FATAL: Error running install command for wctdm24xxp
I think i have successfully
Actually have downloaded a database of over 3000 US numbers in it so
far, and have a script to return an XML result with data in it. Coding
of the page to enter data is also on the way and should be up and
working soon. We are still working on adding the names to the numbers
that we do have,
So a few questions...
1. Can we dump the entire database with names, category and the
collective hit count/sync anything we already have.
2. Can we upload our hits to the collective count (and immediately 0
ours, reloading our collective count number) on a periodic basis.
3. Can we upload new
3. a list of bogus entries..so when you look at it, you know it's a
fake phone number...one that recently came in that got me thinking
this was 407 111 .
I don't know much about the legal position over the other side of the pond, but
I'm pretty sure that in the UK caller ID spoofing is
Suspect it is however I get one spoof a day on average. Most I recognize as
invalid dialplan numbers though
There's a thought, a incoming dialplan validity checker... If a number can't
pass a basic nxx-nxx-, or is from an unassigned npa, auto zap it
That said, incoming skype are often
Hi,
Could anyone tell me how to read the values in the zonedata.c file? I am
looking at the zt_tone_ringtone field mainly.
Thank you.
Jad Wauthier
___
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asterisk-users mailing list
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Hi everyone,
Is it possible to call from your Asterisk server to the Skype network?
i.e., let's say I would like to call from an extension from my
Asterisk PBX machine to a Skype account, is this possible?
I did a little bit of searching and they were talking about that's
only possible with
On x86 asterisk systems, there's 3 options out there, of which the
Chanskype one I've found to be the best. It's $20 US for a single
channel personal license or $99 / per channel on a business license. On
the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to
configure it. I've made
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
On 5/12/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Jon Pounder wrote:
On 5/11/07, Alex
I've solved this problem. It was very easy (only if I knew how to do it
before). I changed the UDP ports, i.e.
1. In sip.conf, bindport=5070
2. In my IP Phone server settings, www.myserver.com:5070
Now it seems to be working good and I hope there'll be no more problem with
it.
Brian Capouch wrote:
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted
[EMAIL PROTECTED] wrote:
Stephen Bosch wrote:
Is Marmite also available in Ontario, or only Out West?
As far as I know, Marmite is available all across this land, from sea to
sea to sea.
Three cheers for Marmite.
IMO most Americans have never even *heard* of Marmite, much less tasted
[EMAIL PROTECTED] wrote:
In todays socio/political climate, telco infrastructure is seen as
foundational, and an essential service that is vital in times of
emergency. Any unauthorised modification can present an unacceptable risk
exposure to the telco, the emergency services, and to the
C F wrote:
Stephen i disagree. growing up in new work city i can say its quite
easy to get away with it in the city. where i live now in new jersey
(population of around 6) i wouldnt be able to pull that off.
The world is a big place, and I suppose there's room for all kinds. In
these
Hi,
Pretty sure I'm missing something simple, but I've seen references to
this feature but not found documentation for it:
I have a queue set up so that many people are contacted (ringall) when a
call comes in. I would like the answering party to confirm that he is a
human being rather than
From: Yaakov Menken [EMAIL PROTECTED]
Date: Sun, 13 May 2007 00:59:54 -0400
Hi,
Pretty sure I'm missing something simple, but I've seen references to this
feature but not found documentation for it:
I have a queue set up so that many people are contacted (ringall) when a
call comes in. I
So Steven, did the echo problem stopped once the Hardware echo cancellor card
was installed out of the box, or you needed to do some configuration changes
like Rx Tx etc.
Thanks for sharing your experience.
--
Deepak
»Steven Ringwald« [EMAIL PROTECTED] wrote:
Deepak Naidu
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