On Sun, 15 Nov 2009, Leif Madsen wrote:
However, changing the label is probably not really the right way to go
about this. For example, I have created an Asterisk system for a call
centre that uses hot desking with Polycom phones, and those phones then
use the built in web browser with
Hello,
I have an AGI (in C) on 1.4.26.3 that puts a caller on hold, does a few
things, then blind transfers the call (with EXEC Dial...) to a parking
space. This is working fine.
Now I want to add an overhead page AFTER the transfer has happened,
basically announcing that there is a caller
Next question will be How can I keep my server from crashing? :)
(add more RAM... which may have been a good answer for question 1...)
j
On Tue, 24 Nov 2009, Alex Balashov wrote:
Disable swap space.
swapoff -a
Jerry Geis wrote:
Is there a way to keep asterisk in RAM
and tell linux not
On Tue, 24 Nov 2009, Richard Kenner wrote:
On a closely related note, has anyone built a normal (not embedded)
system on SSD?
I've been running Asterisk on a 20GB SSD drive for a while now.
What mft/model?
I was recently quoted a 4GB Compact Flash drive as part of a small system
we plan
On Tue, 24 Nov 2009, Eric Chamberlain wrote:
On Nov 24, 2009, at 6:17 AM, Tilghman Lesher wrote:
Sounds like your local DNS resolver isn't answering queries promptly.
Thanks, I'll look into it. Our CURL function only calls one hostname over
and over.
Would setting CURLOPT
Hi,
Happy Thanksgiving to those of us in the USA...
Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to
this snippet of dialplan:
exten = 00,1,DeadAGI(pq.agi,50)
pq.agi then plays a prompt (which I hear just fine):
[Nov 26 02:42:47] VERBOSE[28721] logger.c:
Try IPComms.
j
On Fri, 27 Nov 2009, Marco Cordeiro wrote:
Hello All,
Do you guys suggest any 1800 DID Provider in the US ?
I'm having a hard time to find one.
Thanks,
Marco
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Slightly OT?
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
To date, broadband Internet connections at both offices have been used
as the link, with a VPN tunnel, and phones in
Hi David,
On Tue, 8 Dec 2009, David Gibbons wrote:
snip
A client has two offices in the Virgin Islands that MUST maintain data
connectivity, and there are no available leased line options to run
a P2P link between them.
snip
Is there line of sight? I've been wanting to do a long-shot wifi
On Fri, 11 Dec 2009, Joseph wrote:
On 12/11/09 14:05, Jonathan Thurman wrote:
On Fri, Dec 11, 2009 at 12:44 PM, Connor Spiess cspi...@idea-ma.com wrote:
Joseph
You could also check out the Audio Codes gateways if the Grandstream
doesn't work out for you. They make FXO/FXS
gateways. They
On Tue, 15 Dec 2009, Ben Schorr wrote:
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones.
I've got G.729 loaded in the modules on the Asterisk server and on the
Polycom phones I've set G.729 to be the first preference of codec, but
still when I go SIP SHOW CHANNELS during active calls
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI
On Wed, 6 Jan 2010, Arun Sasidhar wrote:
Hi,
I dont know the type of caller ID. What you mean by this?. I am from
India. I don't know more about this.
*
Thanks,
Arun S*
Hi Arun,
Just for fun I read over the bug id you quoted below, and it seems there
are a number of settings you may
On Tue, 12 Jan 2010, Danny Nicholas wrote:
Since you are small, trixbox would probably be the ideal flavor of Asterisk
for you. It is a downloadable ISO that installs Scientific Linux and
Asterisk and sets you up to manage everything with a GUI interface from a
browser. Once you outgrow
On Tue, 12 Jan 2010, Richard Kenner wrote:
And, I'd be in the camp that would advocate getting your hands dirty and
learn to program without the GUI. You'll learn a lot and then if you'd
want to move to a GUI and something breaks, you'll have some idea on
what and how to fix it.
Knowing
On Tue, 12 Jan 2010, Richard Kenner wrote:
Your comments both come from having taken a short look at FreePBX and
dismissed it without investigating how powerful it can be.
Yes, but the discussion is about COMPLEXITY, not power!
I thought the discussion was about how an IT guy with no
On Thu, 14 Jan 2010, Doug Lytle wrote:
Kingsley Tart wrote:
Hi,
I'm trying to receive faxes using hylafax / iaxmodem but I just can't
get it to work. We're using Sangoma E1 cards and have calls coming in
Without seeing your config files for iaxmodem and hylafax and also
seeing a
Hello,
My first attempt to get dahdi running on 1.4.28... with a Rhino 8 port
modular card and a single FXS module.
Got the Rhino card installed and the machine sees it:
r...@pbx:/etc/dahdi# dmesg | grep rcbfx
[ 71.985309] rcbfx :04:00.0: PCI INT A - GSI 21 (level, low) - IRQ
21
[
there is no change.
Still no channels to show in asterisk. Interesting that it just randomly
decided to create extensions 4000 and 4001 for my two channels :)
j
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
Perhaps this more belongs on the FreePBX list, but for the archives, this
is what I did to make it work:
chan_dahdi wants to read /etc/asterisk/chan_dahdi.conf
FreePBX, at least how I installed from source, seems to think I am still
running Zaptel. It created zapata_additional.conf when I
On Thu, 14 Jan 2010, Julian Lyndon-Smith wrote:
Has anyone managed to get these two phones to make a video call to each other
?
If so, care to share how the hell you managed ?
the GXV is at the latest firmware, and xlite the latest download
Asterisk 1.4 trunk
TIA
Julian
I've done
On Fri, 15 Jan 2010, randall wrote:
Sure. My point was just that IF you only got one connection in the wall,
its cheaper to get a switch than getting a phone with dual 1Gbit ports.
Leif
OK, point taken.
but i have 6xisdn2 and already 2x24 gigabit switches (will need to replace
one
On Fri, 15 Jan 2010, Hans Witvliet wrote:
If you connect your pc with GB-lan card to an dual-ported ip-phone, you
and up with an 100Mbps lan connection to your pc.
Only way to avoid that, is to insert a cheap second lan-card in your pc,
and connect your phone to the second lan, so your pc
On Thu, 21 Jan 2010, Gergo Csibra wrote:
Wednesday, January 20, 2010, 11:41:48 PM, Michiel wrote:
Forget about virtualization!
...
Virtualisation is nice for test-setups, but thats it. for any real job
it's a major pain in the ass and makes stuff bork beyond imagination.
Well. Why do
Hi,
A potential client (hotel) has a Property Management System that talks the
Mitel protocol to their current Mitel PBX in order to receive CDRs
(which end up being rated by the PMS system and charged back to guests).
Does anyone know of any (free or otherwise) docs on this protocol, or
Howes said:
On 27 Jan 2010, at 15:48, Jeff LaCoursiere wrote:
Sounds good to me, but without the spec I'm stuck in a catch 22!
tcpdump? (assuming IP). Bet its fairly simple plain text or something.
Steve
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On Tue, 2 Feb 2010, Steve Edwards wrote:
On Tue, 2 Feb 2010, wassim darwich wrote:
Thanks for?your reply,ill give?you my situation, iam using my asterisk box
as a switch ,so my client is sending me ulaw and my voip provider?only
accept g723 ,So what i have to do is to receive?g711?codec
Has anyone played with the idea of Asterisk as an H.264 multicast tool?
I am wondering what the possibility would be to have some kind of machine
with a capture card call asterisk over SIP and have asterisk make another
hundred calls to subscribers. Then any H.264 compatible device (Android?
On Fri, 5 Feb 2010, Nikhil Nair wrote:
Hi again,
OK, I've now installed a local caching nameserver, but don't see any
change at all.
IN detail, what I did:
- Installed Debian packages resolvconf and dnsmasq (resolvconf just takes
care of dynamic nameserver allocations in
On Fri, 5 Feb 2010, Vinícius Fontes wrote:
I solved similar issues by setting srvlookup=no, having bind running
locally and just the line nameserver 127.0.0.1 on /etc/resolv.conf.
Your local bind is what solved the problem. The srvlookup=no didn't
actually help IMO.
j
On Fri, 5 Feb 2010, Mark Willis wrote:
Has anyone done any large scale intercom deployments with Asterisk? I've
been asked about building a system to one-way page 500 phones
simultaneously from a single server.
My concerns are:
- My limited math capabilities suggest 41 Mbps of RTP
On Sat, 6 Feb 2010, Thomas Perron wrote:
karl,
does it make you feel good ?
wow. pathetic.
On Fri, Feb 5, 2010 at 11:00 PM, Karl Fife karlf...@gmail.com wrote:
Try this:
#rm -rf /
I second that opinion. Tell us first WHY you want to dial 1 numbers
in sequence. Without any reason,
fail.
On Tue, 9 Feb 2010, aster...@opensourcesolution.in wrote:
test
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On Wed, 10 Feb 2010, Tim Nelson wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
If not using PoE I'd suggest getting a few extra PSUs though - that's
one
area I have had a few issues with - but maybe it's just been the UK
ones.
Gordon
The same can be said for the US
Playing around with the Grandstream GXV3140.
I'm interested in having the video voicemail clips emailed in a format
that might be opened by Windows Media Player or even Quicktime. Have been
googling around a lot and have tried various bits of OSS to read the
resulting .h264 file that
On Mon, 15 Feb 2010, Tilghman Lesher wrote:
On Monday 15 February 2010 14:09:38 Olle E. Johansson wrote:
15 feb 2010 kl. 20.31 skrev Jeff LaCoursiere:
Playing around with the Grandstream GXV3140.
I'm interested in having the video voicemail clips emailed in a format
that might be opened
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send over SIP?
Cheers,
j
--
On Thu, 4 Mar 2010, Dave Fullerton wrote:
Jeff LaCoursiere wrote:
I had a customer ask me about time/date information being sent to his
analog (attached to a Linksys SPA2102) answering machine. I didn't know
that POTS could carry this information. Is this something Asterisk could
send
On Thu, 4 Mar 2010, Steve Howes wrote:
On 4 Mar 2010, at 23:11, Steve Edwards wrote:
On Thu, 4 Mar 2010, Steve Edwards wrote:
On Fri, 5 Mar 2010, David @ULC wrote:
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
Record
On Fri, 12 Mar 2010, Angelito Manansala wrote:
If you are having trouble reading this email, read the online
versionhttp://now.eloqua.com/es.asp?s=491e=78675elq=55426a8b6c714f5bb6f2bf4b5d37bf55
.
http://app.en25.com/e/er.aspx?s=491lid=215elq=55426a8b6c714f5bb6f2bf4b5d37bf55
Dear Lito,
What does the (T) mean? Am playing around with running an IAX trunk over
an OpenVPN session and see this only on this peer.
demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK
(26 ms)
Same thing on the other side:
sunfone/demopbx 10.222.0.1 (S) 255.255.255.255
On Sat, 13 Mar 2010, Jeff LaCoursiere wrote:
What does the (T) mean? Am playing around with running an IAX trunk over
an OpenVPN session and see this only on this peer.
demopbx/sunfone 10.222.0.6 (D) 255.255.255.255 4569 (T) OK
(26 ms)
Same thing on the other side
On Mon, 15 Mar 2010, Ishfaq Malik wrote:
Conrad Wood wrote:
FWIW, just received an android-based phone and after installing
sipdroid found that it works very well with asterisk ;).
It automatically dials numbers through asterisk if available and
otherwise through the gsm network.
On Mon, 15 Mar 2010, David Backeberg wrote:
and also to do LCR and Quality based routing of International calls?
I don't know what that means.
Least Cost Routing. Asterisk doesn't have anything built in for this. We
do it with an in-house AGI. Others have done similar things that you
...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Monday, March 15, 2010 6:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways
On Mon, 15 Mar 2010, David Backeberg wrote:
and also to do LCR
On Tue, 16 Mar 2010, Pham Quy wrote:
Hi all,
This question has been asked for days, I think that would be more
comprehensible if i post it in a new thread.
What i want to do is something like karaoke. when users call to
asterisk, a music song is played while caller sings. Their voice
On Sat, 20 Mar 2010, Loic Didelot wrote:
Hi,
I try to get an 8 Port Junghanns BRI card working under dahdi. The card
works with zaptel but I have no success under dahdi.
I load the module with modprobe wcb4xxp. I dont get any errors but I
dont see the spans in /proc/dahdi. The output from
I finally got it at Calories Consumed. Geesh. Good one! :)
j
On Thu, 1 Apr 2010, Olle E. Johansson wrote:
FOR IMMEDIATE RELEASE
Puerto Escondido, Mexico, April 1st, 2010:
Digium launches Asterisk VCC (TM) - a new virtual communication platform
for enterprises, the public sector and the
Howdy,
Can anyone point me to links or discussions about realtime jitter
measurement? I read a long thread from 2007 (Douglas Garstang) that
didn't end with any conclusions. I want to do the same thing he was
trying to do - allow realtime jitter measurements to help control call
routing with
On Thu, 8 Apr 2010, bruce bruce wrote:
I am not sure if unplugging line from card would work as it's still in a
hunt and calls will keep coming through that number and won't fall over to
next line unless there is a BUSY on the line. There is no timeout; it's a
hunt on BUSY. Plus, I don't
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central location,
then have clients that connect via SIP to us for access to those trunks.
Most of them are just fine, but lately we have a handful that are
On Thu, 8 Apr 2010, Tim Nelson wrote:
- Jeff LaCoursiere j...@jeff.net wrote:
What is the consensus on using the 1.4 jitterbuffer? Do most people
enable it?
We have a PSTN server that has our RBS T1 trunks in a central
location,
then have clients that connect via SIP to us
On Fri, 16 Apr 2010, Carlos Chavez wrote:
On Fri, 2010-04-16 at 08:38 -0700, Luki wrote:
Please note: A Zaptel timer must be present for conferencing to work!, but
if the user does not have ZAP/DAHDI hardware, he can use ZAP/DAHDI DUMMY
Actually, my understanding is that this is incorrect.
On Fri, 16 Apr 2010, Nathan Clemons wrote:
I'm looking to find a test tool that will register with our Asterisk
(Trixbox) server here at work and place an outgoing call via our main SIP
trunk (BroadVoice) to confirm that things are working. I've looked around
but I can't seem to find any
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay. I'm
seeing prices around $45 to $50 for this
On Thu, 6 May 2010, Sebastian Milioto wrote:
It is a building, with 24 separated rooms, each room will have a PC and a IP
Phone. Every room connected to a switch Cisco 2950.
I want keeping all PCs isolated behind a NAT (no access to neighbour's PC),
and still keep communication in same LAN
On Thu, 6 May 2010, Sebastian Milioto wrote:
I see the following in SPA922 System tab (new firmware)
VLAN Settings Enable VLAN:yesnoEnable CDP:yesno VLAN ID:PC Port VLAN Highest
Priority:01234567No Limit Enable PC Port VLAN Tagging:yesnoPC Port VLAN ID:
VLAN ID:1 for all Phones, and VLAN
I'm pretty sure you want it to say naught to make the british happy, for
zero anyway...
j
On Tue, 11 May 2010, David Backeberg wrote:
Make it say 'zed'.
It will make the British happy, and cause a different kind of
confusion for the Americans.
On Tue, May 11, 2010 at 4:09 PM, Richard
On Thu, 20 May 2010, Gordon Henderson wrote:
On Thu, 20 May 2010, SIP wrote:
Even IF you could get a keyboard with lights you could individually turn
on and off (never seen one),
http://www.artlebedev.com/everything/optimus/
Bit expensive though...
Gordon
Heh. A $2400 keyboard.
Something new to me. Recently installed a 1.4.30 box for a small office
with four POTS lines in a hunt (Digium TDM410P). Had the telco put a
call forward option on the main line of the hunt. They dial a feature
code from their desk phones (Polycom IP450) that results in forwarding the
main
I have several Atom based boxes running OpenVPN and processing up to six
simultaneous calls over it with no issues. I am quite sure it could do
more. Load is still at .2 :)
j
On Wed, 26 May 2010, Andrew Hakman wrote:
I use openvpn for VOIP traffic all the time. It's not a commercial
rings
or seconds before answering, that would be great. I'm coming up zero on
searches. Its already set to wait for callerid, so I am a bit confused
why it is picking up on a splash... seems it should wait for that second
ring anyway.
Cheers,
j
On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote
On Thu, 27 May 2010, Mike wrote:
Hi,
I have a test server with 2 NICs, each with it own IP address. Let`s say
192.168.1.2 and 192.168.1.3. I would like some phones to register by using
192.168.1.2 and some by using 192.168.1.3 as the address.
Since the default IP is 192.168.1.2,
On Tue, 1 Jun 2010, Mike wrote:
Thanks Joe,
They are on different segments. Those two NICs share nothing but the
server.
But more to the point, it doesn't explain why a simple routing rule matching
the destination by IP address works wonderfully, but not one where I match a
fwmark
We have been distributing asterisk servers for several years now, and early on
decided that hardware echo can was the way to go. Our first few boxes without
it had horrid echo problems, and attempts at tuning in 2006 didn't make any
difference.
We installed a new server yesterday at a
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system is running 1.4.28 and Dahdi 2.2.1-RC2. I understand
that
when Dahdi detects no HWEC, it enables SWEC
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
On Thu, 10 Jun 2010, Gordon Henderson wrote:
On Thu, 10 Jun 2010, Jeff LaCoursiere wrote:
We are totally out of touch on the subject of software echo cancellation in
asterisk. The system
I don't know how it calculates it, but FreePBX shows a bar for total
calls that looks like it maxes out at six. We haven't hit that on any
installs of this device yet, but that seems pretty low for sure.
I know with four calls in progress, all VoIP, transcoding G711u to G.729,
the load of
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 3:46 AM, Michael Graves mgra...@mstvp.com wrote:
Some distro's, like Askozia and Astlinux, have been specifically
engineered around running from flash media. This basic form of
operation has been well proven in projects like
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be charging a monthly fee that is
disproportionately high compared to my need to let
Hi,
I have several 1.4.29 installations with Sangoma AFT101d cards. Normally
we have been collecting the raw data and then graphing channel use for
these customers with:
asterisk -rx 'show channels' | cut -f1 -d' ' | grep Zap | sort -u | wc -l
Then I recently noticed that there were some
there is no way other than what I am already doing to judge
the channels in use?
Thanks,
j
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent: Tuesday, June 22, 2010 11:41 AM
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
A more overriding problem for me is how do we know what *destinations* to
filter so this idea of war dialing a toll
On Wed, 23 Jun 2010, Gordon Henderson wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
A more overriding
On Wed, 23 Jun 2010, Tarek Sawah wrote:
you can start by simply telling us what is the purpose of your server..
and does it have long distance of overseas?? do you use Numeric
usernames? simple passwords? passwords the same as your username? this
way you can offer more info so we can
On Wed, 23 Jun 2010, Steve Edwards wrote:
On Wed, 23 Jun 2010, Jeff LaCoursiere wrote:
Some !...@$#@@# in the Czech Republic used one of our SIP accounts to place
four thousand calls to what appears to be a toll number in Zimbabwe last
night. Filter 82.150.165.5.
Ouch. 82.0.0.0/8
I am sure this is simple, but have been struggling. I want to create a
call file that dials out a particular Dahdi channel to enable call
forwarding on a POTS line. I have this in extensions.conf:
[custom-callfwd]
exten = s,1,Answer
exten = s,n,Dial(DAHDI/4-1/*717157750)
exten =
On Wed, 30 Jun 2010, Steve Edwards wrote:
Now I whipped up a C program to create a call file to do the same thing
from the command line:
[snip]
fprintf(callfile, Channel: Local/*...@custom-callfwd/n\n);
I don't see exten *71 in custom-callfwd.
Doh! That was the problem. In
On Tue, 6 Jul 2010, C.Savinovich wrote:
I am writing to you privately... [snip]
Doh! Need another cup of coffee?
j
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On Sat, 2010-07-17 at 00:08 +0530, Anita Hall wrote:
Hi
Is it possible to receive video calls using Asterisk and then process
them as an IVR ? One of our clients wants to set-up a video IVR system
in the US and we are evaluation possible options.
Also, what is the bandwidth of receiving
On Mon, 2010-08-02 at 14:26 -0400, bruce bruce wrote:
Hi Everyone,
Sorry, if it's not directly related to Asterisk. Some of people on
this list might have PBX deployed for their clients. What software do
you use to invoice them so the invoice looks like a proper telecom
invoice maybe?
On Tue, 2010-08-03 at 09:17 -0400, bruce bruce wrote:
I agree but the mentioned software is not opensource.
My conditions clearly included opensource.
No, your prefer listed opensource. If you had said requirement I
wouldn't have suggested it.
j
On Tue, Aug 3, 2010 at 12:35 AM, Nick
On Sun, 22 Aug 2010, David Backeberg wrote:
On Sat, Aug 21, 2010 at 10:49 PM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
Voice recognition is a pain for people with accents and poor lines and when
Everybody has an accent. Some people live in a place where the people
they talk to sound
On Sun, 22 Aug 2010, Jason Aarons (US) wrote:
I'm not aware of an open source speech product.
Some great examples where speech recognition works well are
1-800-USA-RAIL, Microsoft/Cisco corporate directory 425-882-8080 you can
say the employees name and be connected and those works
On Mon, 23 Aug 2010, Don Kelly wrote:
I’m looking for a way to use our implementation of HylaFax on Asterisk with
Cardiff (an
old installation of Cardiff document stuff).
Is someone doing that?
If no one has direct experience, is there a HylaFax client that emulates WinFax
print-to-fax?
On Fri, 2010-09-10 at 23:07 -0400, bruce bruce wrote:
Hi Everyone,
I have a provider whose DID used to come into the box just fine but
recently stopped working. Nothing has been changed on our end.
Here is what I get when doing sip set debug peer PROVIDER:
Sending to
On Sat, 2010-09-11 at 03:53 -0400, Zeeshan Zakaria wrote:
This is not elastix or FreePBX forum and asking non-asterisk related
questions here is misusing this mailing list. Allow anonymous sip is
not an asterisk feature. Look in the code in extensions.conf what it
is programmed to do and
--
www.ilovetovoip.com
On 2010-09-11 7:22 PM, Paul Belanger
paul.belan...@polybeacon.com wrote:
On Sat, Sep 11, 2010 at 2:41 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
Sending to 123.123.12...
Either you changed the peer parameters or they did...
If he
[snip]
customers, who all connect from behind their home nat gateways of all
kinds. I still don't know why that fixed it.
Sorry you took it so harshly Zeeshan, but the only posts that stick out
to me from you are the ones where you are bashing people for posting
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have a customer that had us port one of their
800 numbers and apparently didn't realize that they had published that
number in Canada as well. Our current
On Wed, 15 Sep 2010, Kyle Kienapfel wrote:
On Wed, Sep 15, 2010 at 6:04 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Tue, 14 Sep 2010, Joe Freeman wrote:
Anyone have a good provider for International (US/Canada at least) 800
termination/origination? I have
Hi,
I have an asterisk 1.4.35 server with a Digium TE410P (1st gen) four
port T1 card. Only one RBS T1 plugged into it right now.
I have been getting complaints about random hangups. Endpoints are all
remote, but I track very closely the latency (by graphing the output of
sip show peers) which
On Tue, 21 Sep 2010, Shaun Ruffell wrote:
On 09/21/2010 10:48 AM, Jeff LaCoursiere wrote:
I have several servers with Sangoma A104d cards, and the Sangoma driver
has a debug mode that lets me see the RBS bit transitions. I have used
this in the past to prove that the T1 provider is actually
On Fri, 8 Oct 2010, Bryant Zimmerman wrote:
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the
three perform well in all
enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer
some combination of issues.
I am looking at Patton and Innomedia.
Hi,
Embarrassed as I am to write this, I am hoping for some advice. One of
our very first PBX installs, now six years old, was taken advantage of
over the past few weeks. A victim of sipvicious, I assume, that managed
to guess one of the SIP passwords. 4000 calls to various middle eastern
On Thu, 21 Oct 2010, Steve Howes wrote:
Hi,
Given the recent increase in SIP brute force attacks, I've had a little
idea.
The standard scripts that block after X attempts work well to prevent
you actually being compromised, but once you've been 'found' then the
attempts seem to keep
[snipped very confusing top and bottom posting mix]
On Thu, 21 Oct 2010, Sherwood McGowan wrote:
Dhaval,
You're right, I forgot one thing. The frozen table's id column should not
be an autoincrement, it should be set by the insert statement, using the
original method I decsribed for
On Thu, 21 Oct 2010, Andrew Latham wrote:
Always start here... http://www.spamhaus.org/drop/
If the AS is stolen, you can block the network and never have to worry
about it...
~
Andrew lathama Latham
lath...@gmail.com
I guess you are assuming that spam networks should be included in
On Thu, 21 Oct 2010, Steve Howes wrote:
On 21 Oct 2010, at 16:54, Jeff LaCoursiere wrote:
I'll subscribe, that is for sure. What is the best way to dist the
blacklist? iptables include file? Or something more integrated to
asterisk... just thinking off the top of my head that a module
On Sat, 30 Oct 2010, Joel Maslak wrote:
For me, monitoring outbound call volume makes a lot more sense. I would
love to see an easy to use, out of the box method to alert me if more
than x number of erlangs* are exceeded within a five minute, sixty
minute, and one day time period. For
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