We upgraded from asterisk 11 to asterisk 13. Recordings were working fine in
11 but nothing is being written on 13.
Here is the dialplan segment
same =
n,ExecIF($[${TL_PHONE_CALL_RECORD}=TRUE]?SET(CONFBRIDGE(bridge,record_conference)=yes))
same =
signalling = em_w
immediate = no
group = 1
channel = 1-3
Bill Dunn
- Original Message -
From: Ron Bergin
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, July 06, 2012 12:34 PM
Subject: Re: [asterisk-users] DAHDI DTMF problem?
Bill Dunn - VCI Internet
?
Bill Dunn
- Original Message -
From: Shaun Ruffell
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, July 06, 2012 12:16 PM
Subject: Re: [asterisk-users] DAHDI DTMF problem?
On Fri, Jul 06, 2012 at 11:10:43AM -0500, Tim Nelson wrote:
- Original Message
How big is the blocklist from fail2ban? - a few thousand entries and the
network stack performance degrades.
BillK
On Sun, 2011-07-31 at 19:54 -0400, C F wrote:
How long ago was the last block from fail2ban?
What could be is that the attacker hasn't yet realized that he has
been blocked and
fail2ban might be good for this.
On 04/05/2011 01:00 PM, asterisk-users-requ...@lists.digium.com wrote:
Date: Tue, 5 Apr 2011 08:44:41 -0700 (PDT)
From: Steve Edwardsasterisk@sedwards.com
Subject: Re: [asterisk-users] Iptables configuration to handle brute
force registrations?
On
List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc: Bill Michaelsonb...@cosi.com
Message-ID:banlktimqrbfmqpoinrphr_rjekolbwp...@mail.gmail.com
Content-Type: text/plain; charset=iso-8859-1
On Tue, Apr 5, 2011 at 1:31 PM, Bill Michaelsonb...@cosi.com wrote:
fail2ban
On Sun, 2011-02-13 at 22:54 -0800, Steve Edwards wrote:
On Mon, 14 Feb 2011, Bruce B wrote:
What sources do you use to limit SIP connecting customers to specific
countries by IP (e.g. allowing USA and not China). It would help me a
lot of you can note the sources you trust that are
but it seems like an awfully lot of baggage to carry around
just to get the IAX part of the project. Any pointers would be greatly
appreciated.
Best,
Bill
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New
On Tue, 2010-04-27 at 11:01 -0400, John Novack wrote:
Anita Hall wrote:
Hi
Please check out this product
http://www.sangoma.com/products/hardware_products/data_networking/a301.html
Does it work on Asterisk or Freeswitch ?
Do Telcos provide an E3 connection ?
One of our
This is why I don't do this kind of work anymore. Impossible to
distinguish the phantom problems from the real ones - and I'm convinced
there ARE phantom problems when you install new telephones on people's
desks.
Suggestion: learn to use the facility in Wireshark that can log a
SIP/RTP
Keep in mond that cron usually has a very abbreviated environment for
security reasons - you may need to set the PATH or other environment
variables in the crontab to get it to work.
Billk
On Wed, 2009-12-09 at 20:55 -0500, Michelle Dupuis wrote:
Interesting...I'll try that. Thanks
-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bill Kenworthy
Sent: Wednesday, December 09, 2009 9:31 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Can't restart asterisk from script
Keep in mond that cron usually has
Hi All,
When typing 'help' on the command line (* console) is there a way to
keep it from just scrolling most of the information off the top of the
screen? I can't hit ctrl-s fast enough so I miss most of the info. This
makes 'help' be not much help.
Thanks,
Bill
-enter the asterisk cli? Kind of defeats the purpose of
'online help' doesn't it?
Not trying to be a smart-a$$, just hoping to find something a little smoother.
Is there a better way, or is help as useless as it is starting to appear?
Thanks,
Bill
--
Date: Tue, 17
before I get started on it. Does anyone know of a hardware interface
that is already being manufactured that can tie a codec-based device
into Asterisk?
Thanks in advance,
Bill
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Can Asterisk be configured to hang up if another phone picks up?
I'm a bit lost as far as terminology goes, but here's my setup. At
home, I have asterisk answering calls from the pstn and sending them
through to a sip extension or voicemail. All that is working fine.
The box running Asterisk
I get how everything is connected with your setup, but if you pick up
the cordless phone to answer a call does the sip extension just keep
ringing until it times out?
I like the exclusion adapter idea because it sounds like it would let
me keep my dialplan intact. But I do take John and
Polycom IP 430 or 330.
asterisk-users-requ...@lists.digium.com wrote:
Date: Mon, 16 Mar 2009 18:24:33 -0400
From: David Ruggles da...@safedatausa.com
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125
It's conceivable that the combined effort of these two responders
required less than ten minutes of time, yielding a theoretical pay rate
of $120/hour.
I wonder how much effort went into the other responses.
That will be $6 for my commentary, please.
Folks wrote:
Message: 1
Date: Tue, 3 Mar
Is there a convenient way to limit the number of call files (outgoing
directory) that are processed concurrently?
smime.p7s
Description: S/MIME Cryptographic Signature
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What economic downturn?
I'm sick and tired of hearing this mantra.
I wish you the best of luck in maintaining your immunity.
Same here (in the UK).
As long as people need to make phone calls ...
Gordon
The economy (and indeed
David @ULC ucoms2...@gmail.com wrote
One of my user was asking, can he use VPN to access asterisk ?
What does it mean ?
And its possible ?
How ?VPN
Sometimes what is called a VPN is not a VPN by everyone's definition, so
beware. By my definition, a (IP) VPN supports full layer 3
How are you getting these 80 or 120mm fans in a 1U chassis? Remember you
got barely 45mm to play
with at the back and front of the switch. How are you going to mount a
80mm or 120mm fan on there? Are you assuming that the units mounted
above (or below) your switch is a short 1U? You can't
such a bad thing?
A: Because it messes up the order in which people normally read text.
In the order in which people normally read text they don't
repeat the entire conversation from the beginning each time
a question is asked either... Bottom posting is just as bad!
./bill
-Original Message
Sometimes I do. It depends on my mood and purpose. And sometimes the
author prefers to write last things first, for whatever reason.
I'm kind of agnostic, too.
Mike Dent wrote:
H, not sure about you but I often pick up a book and flick from
the back to the front, does nobody else do that?
From: Doug [EMAIL PROTECTED]
Net Neutrality is great in principle. But ISP's need to
somehow control those few percentage of users who suck down
a huge majority of the bandwidth. It's dollars and cents.
There is a rational solution for the traffic management issue. It just needs
to be
Jared Smith had written:
To answer the second portion of your question (which I forgot to do in
my earlier email)... yes, Asterisk can be a registration server as well.
--
Jared Smith
Training Manager
Digium, Inc.
Valentin Bud wrote:
Hello Mr. Smith,
snip
If you know any kind of books
On Sat, Nov 15, 2008 at 03:49:40PM -0700, Joseph wrote:
What kind of IAX2 client will install/run on EEE PC 1000 (stock Linux
software)?
I'll eventually replace this crippled Linux with something better but I don't time to play around with it as most divers and modules are still too new
I'm wondering how prevalent the practice of physically segregating voice
and data networks is in the Real World.
What are the factors that typically lead to such a decision?
DIscussions of pros and cons are most welcome by me.
Experiences, anybody?
smime.p7s
Description: S/MIME
Alex Balashov wrote:
Send asterisk-users mailing list submissions to
asterisk-users@lists.digium.com
I'm pretty sure they meant two logical networks. At least, I hope they did.
Unfortunately, I was indeed referring to two physical networks. Cabling,
switches, everything, all the way
back to a POTS
line because no matter what I tried, I couldn't get more than about
80% of my faxes to go through reliably. Of course, we send about
300-400 faxes a day... You cannot reliably send faxes over VoIP.
./bill
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to disable that softkey.
I've looked through the sip.cfg file and can't seem to figure out what
option would remove that softkey. Has anyone ever had to do this?
What feature should I disable?
TIA
Bill
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Kristian Kielhofner wrote:
On 10/23/08, Bruce Komito [EMAIL PROTECTED] wrote:
We've had LOTS of problems with Sonicwalls doing bad things to SIP and RTP
connections. I've seen the delay thing, as well as the Sonicwall throwing
away entries from the ARP table because of inactivity.
be a downside to that
wireless headset :)
Thanks again.
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darryl Dunkin
Sent: Friday, October 24, 2008 3:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] OT: Disable
function.
Bill
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Sorry for asking the obvious question, but are there other elements of
the slow path besides the Sonicwall? I mean, what is in front of the
Sonicwall? Also, might the Sonicwall be positioned as some kind of choke
point in the topology, thus leading to genuine sporadic congestion?
James
I recently sent this email to a user in response to a problem report of
phone calls going to voicemail without the phone ringing. I'm wondering
if I've covered all bases, or whether there is some logical explanation
I haven't considered, and generally what others' opinions/experiences
are
Steve Totaro wrote:
My only wish is that Linux had a facility like XP to bridge NICs without
running all sorts of commands for brctl. Just a GUI like XP. Last time I
setup a bridge in Linux, I had to change many kernel options and rebuild the
entire kernel to get bridging working properly.
Jay R. Ashworth wrote:
In our call center (250 agents), we use Panasonic KX-TCA60 headsets, which
I think we get for 7 or 8 bucks apiece, and sell to the agents at cost.
Thanks for that - they look good, and I found several recommendations for them
after I got yours and started looking for them.
Some users at a new Asterisk installation with Polycom IP330 phones are
complaining about echo with the amplified headsets they used to use with
their Nortel phones. I listened myself, and I here my own voice
annoyingly loudly, and no headset/phone combination of volume control
manipulation
My experience is very limited, but you asked for any perspective, so...
I put an Asterisk with freePBX on a linode server (linode.com), just to
play with it a few months ago. I can say that it worked to the point of
being able to dial out with my Polycom phone on a FiOS connection,
through
From: Joseph L. Casale [EMAIL PROTECTED]
Does anyone have any perspective on how well Asterisk performs and
scales inside a Xen hypervisor environment?
I tried on many different pieces of hardware with various recent Xen
versions and it always had some level of unpredictability and was not
That is my position, and I appreciate the affirmation, as well as the
offer to determine the carrier. I might email you about that. But having
no business relationship with the other carrier, it is at best awkward
for me to initiate contact on this matter, and this should be obvious to
Interesting to see it done. Vitelity claims it is impossible. The number
is 212-651-5632.
BTW, if you provide the originating number, the underlying carrier can
be determined, either by the pooling or NANPA block it is assigned to,
or its LRN if ported. If you want, you can privately e-mail
I have a Vitelity DID which generally works, but calls from a particular
caller do not reach it. Vitelity has thus far disavowed any
responsibility for working through this problem. I recognize that some
action might be required by another provider which is outside Vitelity's
control, but it
I am preparing to connect an asterisk box with a redfone fonebridge to a
T1 service provider. I am doing this by testing first with another
asterisk and a Sangoma card playing the role of telco.
I formerly had this test configuration operating flawlessly as a PRI
connection. But I
Tzafrir Cohen wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go into service. This is good for a
couple of reasons.
Also note that Zaptel will anyway reserve all the 24 (for T1) or 31 (for
E1) Zaptel channels for the span. So
On Tue, Sep 9, 2008 at 7:17 AM, Bill Michaelson [EMAIL PROTECTED] wrote:
I'm faced with an installation at a client site with supposed PRI service on
a fractional T1.
Steve Totaro wrote:
I usually configure the entire span of 24 channels (23 B + 1 D) and
only the turned up channels go
Can someone suggest a better way of getting rid of these channels? My
solution
so far has been to restart Asterisk... not a good solution.
Thanks
Bill
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AstriCon 2008
I expected to find th module ztd-ethmf[.c...] in support of the redfone
TDMoE product in my zaptel distro (I have 1.4.11). But it's not there.
I am awaiting a response to a trouble ticket from redfone. Can anyone
give me a jumpstart? I can't seem to google this up.
smime.p7s
Description:
. (i.e. Yes, you'll
see
a big difference, maybe a little better or couldn't really see
that
big of a difference)
Thanks for your input.
Bill
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AstriCon
is the features ('show features') and then
try to record a call (*1) and see what the terminal says...
Bill Michaelson wrote:
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk 1.4.x and
FreePBX 2.4+. I'm
My client needs call recording features and would like to initiate the
process in-call (typically *1). I'm installing Asterisk 1.4.x and
FreePBX 2.4+. I'm using Polycom phones. I can't make it work. Would
somebody please give a checklist of items for me to compare my list
against - in the
In case this is useful to others, a tip...
I moved one of my Polycom 501's off it's subnet to another one (I've got
an ether bridge glued to the back of the phone and a wireless card in
the * box acting as AP). Now it is still served by the same Asterisk
box, albeit through another ethernet
I appreciate your report (below), but it's a strange and disturbing coincidence
for me. DTMF out through Vitelity was not working for me until 1-2 days ago
when I changed it from rfc2833 to inband!
Maybe I just missed the change date and I should change it back?
Date: Tue, 22 Jul 2008
Message: 1
Date: Fri, 18 Jul 2008 20:35:47 -0700
From: Dave Platt [EMAIL PROTECTED]
I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is
After much checking and puzzling, I cannot get my Polycom 601 to toggle
call recording with my Asterisk 1.4.21.1.
Via FreePBX, I can set a user to always record, and the recording will
show up in /var/spool/asterisk/monitor.
But if I try to start recording by toggling in-call, no luck.
I
A followup to my own inquiry...
pig*CLI feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# #
Attended Transfer
One Touch Monitor
OK, I had broken the feature.conf fileset, but I just fixed it. Now I
can confirm:
pig*CLI feature show
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer
I'm preparing for a client install of * by doing a fresh one in-house.
Unlike my earlier installation that runs asterisk as superuser, my
current experimental box runs without such privilege. This is causing
it to moan that it can't set TOS. I absolutely don't want to install it
on the
This appears to be a SIP gateway to four FXO ports for ~$250. Has
anybody used it with Asterisk? Comments?
http://www.ipphoneshack.com/products/D_Link_DVG_3104MS_VoiceCenter_4_Port_PSTN_Gateway-193-12.html
Any good reason to pay for a Mediatrix 1204 or some other box instead?
smime.p7s
I'm looking for reports of recent experience with redfone fonebridge2
(with echo can) TDMoE gizmos.
Anybody? Good? Bad?
smime.p7s
Description: S/MIME Cryptographic Signature
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I'm considering using a PoE switch like this...
http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=3023334CatId=2800
...to power as many as 24 Polycom phones of varied kinds.
The sales lit indicates 190 watts available for PoE devices. But I'm
concerned about a
!
Everybody needs to learn the other side before getting involved.
Bill
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IT started a
Page All (which is nothing but a meetme). All of the other phones
would send their status for the Buddy List to the IP601 and would
simply overwhelm it - causing a reboot.
Just a shot in the dark...
Bill
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*1.4
Sorry for a dumb question, but I'm working with my SIP
provider on a problem and I can't answer this question
for them. They don't know Asterisk.
When I do a sip show channels
What is the User/ANR field?
Bill
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to
install.
I don't care how well it works once installed, if I have to spend
hours just figuring out HOW to get it installed, I won't waste
my time. Covide might be a good project, but I KNOW Suger-CRM is
because I set it up in about 15 minutes and SAW that it was...
Bill
was
that if they want people to try their project, they need to make
it easy to try. Or people will go elsewhere.
Nuf said. Sorry. Very OT.
Bill
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Tilghman Lesher
I think you just missed the point of open source. Projects are almost
always I made this to satisfy a need for myself, and it's open for
others
to examine and contribute. If you see a need for an easy installation
process, then by all means, you should contribute that.
Oh,
show as 2 processors in Linux, not
4)
Bill
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Bob G wrote:
Why the guy asked a question?
Yes. But the question was about Drag and Drop transfer applications for
Asterisk.
Can 1EZphone do that? If not, your SPAMMING the list!
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with the 650 over the 601.
HTH
Bill
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Is anyone on the list reselling (or just using) EvolutionPBX
from Intuitive Voice Technologies?? If so, please contact me
off list. Thanks.
bill at mwdental.com
Bill
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, it works, you just don't hear the
audio when it tells you the number - it's a known issue!
Hmmm. That would have been helpful information before I did the
upgrade now wouldn't it?
Bill
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf
Isn't this channel specific? Why is this being added? It does not work
with SIP. It does not appear to be architecturally generic.
This gets added, but yet a channel specific enhancement for SIP that would
be beneficial for endusers does not get added. Again, Asterisk is good at
transferring
On Mon, Mar 24, 2008 at 5:21 PM, Watkins, Bradley
[EMAIL PROTECTED] wrote:
Being able to pass variables around between systems is by *definition*
channel-specific, since the channel driver is the module responsible for
speaking a given protocol. Besdies, SIP already has (and has had for
On Sat, Mar 22, 2008 at 7:17 AM, Philipp Kempgen [EMAIL PROTECTED]
wrote:
http://bugs.digium.com/view.php?id=5014
The response on that issue from Russell is the kind of response that
really ticks me off. No, no, no, we don't want any real features that users
want, we want basic, boring
!
Am I expecting too much?
Bill
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class machine and haven't really had any issues
I feel were related to hardware. Alghough, I agree good hardware is
the key to hardware stability.
Thanks for the comments.
Bill
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with Asterisk, it looks like I really need
to learn the ins-and-outs of Asterisk and ditch my pre-packaged
solution. Off to Amazon for to find TFOT (I want the hard copy :)
Bill
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I just replaced an IP 601 with a new IP 650. We have 2 expansion
modules attached. The lights on the expansion modules light up if
a users gets an INBOUND DID call, but the lights don't light up if
the user makes an OUTBOUND call.
Sip: 2.1.1.0052
Has anyone seen this?
Bill
Actually, UNIX [tm] Describes meeting a standard, and not development
history.
http://en.wikipedia.org/wiki/Unix#Branding
Absolutely! Which is why I referred to Linux as Unix-like and not UNIX.
Linux is NOT licensed to use UNIX(r) per The Open Group's specs.
BSD and Mac OS X are licensed
Alex Balashov wrote:
Linux is UNIX, for intents and purposes related to Asterisk.
Well... not so much! If you want real UNIX, go for a BSD or
God forbid, SCO OpenServer.
Their pedigree is from ATT UNIX (SYS V Rel 4?) which is considered
to be the real UNIX. However, as time has gone by,
every time.
We're getting a 650 in to see if that will fix the problem (as it did for
others)
Thanks
Bill
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that, but we are putting one on
next Monday. We'll see.
Our next plan is to get a 650 and see if it can handle the traffic.
Bill
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of JR Richardson
Sent: Friday, February 29, 2008 9:17 AM
for the offer, I think I've got it figured out using winsock.
Thanks again.
Bill
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., but I'm sure there is a better way
to do this without using Telnet (maybe not?). Any suggestions?
I want to write a presence monitor (a virtual sidecar if you will)
Bill
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asterisk
Wow! That is a good question. I can't believe no one has ever asked that
before.
Seriously, before letting someone join this mailing list, it should ask them
a simple question:
When you don't know the answer to something, do you:
a) blindly ask hundreds of people with no regard to whether it
On Jan 2, 2008 12:23 AM, Daniel [EMAIL PROTECTED] wrote:
Hi there,
is there any howto how do i configure a asterisk/trixbox for mail2fax?
The fax must be send over sipgate or other SIP peers. (i dont have
any normal telephones connected).
Do people even read the mail list anymore, or do
Philip Prindeville wrote:
So I'd venture to say that by August, the Internet will really be *30*
years old.
As Al Gore was born in 1948, I can see that the Internet could be as old
as 30, but not much more. 35 years ago would put him at 25 years old.
And inventing the whole Internet at 25 is
On Dec 2, 2007 3:42 PM, Carlos Rojas [EMAIL PROTECTED] wrote:
I'm looking the softswitch in digium website, anyone test the softswitch?
Nope. No one has tested it or used it. Try the one at cisco.com.
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that it's a true performance issue. We
have tried multiple codex (ulaw, alaw, speex, ilbc, everything 3CX
supports) and have the same issues.
Is there some step I'm missing when using a multi-proc machine?
Thanks
Bill
PS: I have been trying to get this to post to the list for a while and
am
2216
Server Chassis: Tyan GT24 http://tinyurl.com/2b5l3r
I was thinking that I was getting timing issues and that perhaps using
ztdummy (and timing_internal?) would help.
Bill
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I use a mysql script to dynamically generate the page command
and page about 70 phones, and I have never had a reboot problem.
Sometimes there is a slight delay waiting for all the phones to
join the page conference. I am using a mix of 650's, 550's, and
330's.
It must only be an issue if
side cars. We have to keep presence (Buddy List) enabled so
the sidecar lights go on and off. However, about 1 out of 10 times
the receptionist pages, her phone reboots. Polycom says it can't
handle the traffic from the buddy list presence notifications.
Have you seen this?
Bill
).
If one begins to fail then you can switch over the periodic back up and
used a new device for backup.
Bill
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan
Sandro
Sent: 12 September 2007 13:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
originated from the list (with my subscribed address).
Make sure your NEW posts are sent from the subscribed address...
Bill
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to my queries and don't keep showing me all the
current call status? (less verbose?)
Bill
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Gordon Henderson wrote:
Either start asterisk with no -v's or type:
set verbose 0
at the prompt.
Thanks. Exactly what I needed.
Bill
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