when postings to the (UGH)
forum happen though.
John Novack
--
Dog is my Co-Pilot
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https
ure email.
>
> --- Original Message ---
> On Wednesday, November 8th, 2023 at 1:21, John Harragin <
> jharra...@mw.k12.ny.us> wrote:
>
>
> > Marek,
> >
> > See if calls hang in the system if you encounter another outage
> > core show channels
>
Marek,
See if calls hang in the system if you encounter another outage
core show channels
...if so,
core set verbose 3
and see what instructions subsequent calls hang on.
On Mon, Nov 6, 2023 at 4:44 PM Marek Greško wrote:
>
> Hello,
>
> sure I have local DNS server and public resolving
)
#--
# Description:
# Author: John Harragin Monroe-Woodbury CSD
# Created at: Thu Nov 6 12:27:35 EST 2008
#
# Copyright: None. Modify and use however you like...
#
#--
# Configure section:
BASEDIR=/var/spool/asterisk
to Federico's issue.
On Thu, Aug 17, 2023 at 5:37 PM C. Maj wrote:
> On 8/17/23 12:44, John Harragin wrote:
> > You should be able to define multiple data sources. However I'm having my
> > own issues. I have my dialplan accessing one maria database which is
> hosted
>
You should be able to define multiple data sources. However I'm having my
own issues. I have my dialplan accessing one maria database which is hosted
locally on the asterisk server then logging cdr with odbc adaptive which
connects to maria on a remote machine. This works fine except when the
using version from git
5c840cf43838e0690873e73409491c392333b3b8 .
So, the question, how to fix, so I can get the tompile to work?
Thanks in advance for any suggestions.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
it!
I have used it as my PSTN provider for more than 10 years, with only one
hacking issue with voip.ms, which they fixed fairly quickly. I see no reason to
change to a protocol that ( it seems )
every thief in the world is banging away on 24/7!!
JMO
John Novack
On 5/27/2023 10:23 AM, Steve
https://github.com/asterisk/asterisk/blob/master/LICENSE#L48 broken
(PS I hope I never find a bug to report, because I don't use Github...
embrace, extend, extinguish is still alive and well)
--
_
-- Bandwidth and Colocation
iaDB ODBC Connector
Anthony,
...anyway, enough about my problems. Have you put a:
Verbose(0, Your built out sql statement)
...before your ODBC application in both contexts to see if you just have
maybe an undefined variable creating a syntax error in your sql?
John
Here is a bit about odbc thr
If there are multiple connections that the utilize the same driver, try
putting:
Threading = 2
in the appropriate driver section of
/etc/odbcinst.ini
...this would be a possibility if the problem is intermittent.
Also can you successfully execute the same SQL from the cli?
By the way,
If you clone one of their repo's you can see their email address in the
commit log...
On Tue, 7 Feb 2023 at 16:56, Jeff LaCoursiere wrote:
> Hi all,
>
> Curious if the github user "mlan" is on this list? Could you please
> contact me off list if so, I was hoping to reference your work in a
You have posted the same message several times in the last few days!!
I would assume no one has an answer to your question, at least on this list.
It seems most have migrated to another (UGH!) venue, so the few that are left
can't help.
JMO
John Novack
Antony Stone wrote:
Hi.
I have
I have had a similar problem. I think geolocation introduced some
additional prerequisites run:
/usr/src/asterisk-X/contrib/scripts/install_prereq test
then recompile asterisk
That script installs a bunch of crap you don't need, but running it in
test mode rather than install might help you
I had similar issues. It looks like modules related to pjsip
(geolocation?) introduced new prerequisites. There is a script in the
source that prepares for an asterisk build. Try running that, then
recompile asterisk and see if that fixes things.
John
On Fri, Dec 2, 2022 at 3:36 PM Justin Piszcz
Trivial issue.
I have a script to rebuild asterisk with the following line:
menuselect/menuselect --disable MENUSELECT_MOH --disable
CORE-SOUNDS-EN-GSM --enable CORE-SOUNDS-EN-WAV --enable app_macro
--enable codec_opus --enable chan_phone --enable
chan_sip --enable chan_sip --enable chan_sip
risk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Your life is like a
have been using meetme and there you can just display the list of
users and you get that information.
Thanks in advance for any suggestions.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
user joins first, it's met the criteria and will conference
> and start recording.
>
> Dan
>
> -Original Message-
> From: asterisk-users On Behalf Of
> John Covici
> Sent: Tuesday, June 28, 2022 6:28 PM
> To: asterisk-users@lists.digium.com
> Subject: [Extern
On Tue, 28 Jun 2022 19:54:11 -0400,
Joshua C. Colp wrote:
>
> [1 ]
> On Tue, Jun 28, 2022 at 8:28 PM John Covici wrote:
>
> > Hi. I have been using meetme for years, but I wanted to try
> > confbridge as meetme is going away soon.I am having a few
>
you spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org
t; Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digi
a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new
with the chan_sip, meetme or something else
entirely? Nothing relevant in the logs.
Thanks in advance for any suggestions.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
?
> >
> > Cheers,
> > Kingsley.
> >
> > On Wed, 2022-01-26 at 16:56 -0500, Dovid Bender wrote:
> > > I tried but it seems it does not.
> > >
> > >
> > > On Tue, Jan 18, 2022 at 2:57 PM John Runyon
> > > wrote:
> > > >
${SPRINTF(%c,38)}
or
%26
should work, I think.
On Sun, 16 Jan 2022 at 13:21, Dovid Bender wrote:
> Hi,
>
> I am trying to play a sound file from AWS S3. The URL is something like
> this http://example.org?foo=bar=b. The issue seems to be that as soon
> as Asterisk see's the & it assumes there
OK, that tells me something, I will disable pjsit for now, learn about
it and try again.
On Sun, 09 Jan 2022 06:39:55 -0500,
John Harragin wrote:
>
> [1 ]
> [1.1 ]
> You can also set up multiple physical or vlan(ed) interfaces and bind sip
> to one and pjsip to the other - then
On Sat, 08 Jan 2022 19:17:57 -0500,
Antony Stone wrote:
>
> On Sunday 09 January 2022 at 00:50:27, John Covici wrote:
>
> > Hi. I am using asterisk 18.3 and freepbx.
>
> Hm, which version of FreePBX uses Asterisk 18.3?
>
> > How can both sip and pjsip be list
ould like pjsit not to listen,till I figure out how to configure
the thing, so my logs don't fill up with messages.
Thanks in advance for any suggestions.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@cc
; Check out the new Asterisk community forum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lis
is a gigantic bit of xml in cisco's
provisioning manual.
Cisco IP Phone 8800 Series and Cisco IP Conference Phone 8832 Multiplatform
Phones Provisioning Guide
I'm hoping I can maintain more minimal configurations.
John (sorry if there are multiples. I just changed my account
. The
Local telco has the DID but the LD does not so I have to verify the DIDs
with the Voip provider(s).
Another case may be for least cost routing.
There are other reasons but you can see that it is not always as simple
as using the same provider for DID and origination.
Thanks,
John
On 3/11/21 3:34
Sangoma purchased Digium.
You can find Sangoma cards at https://www.sangoma.com/telephony-cards/
On Tue, Jan 12, 2021 at 2:29 PM bilal ghayyad wrote:
> Hello All;
>
> We were using Digium cards, now I am not able to reach for digium website
> that contains the telephony cards and Asterisk
phone. I also read that I can google
search for SIP firmware and download them?
Is Cisco 7960 better and more advanced than Cisco 7940?
On 2020-12-18 22:41, John Novack wrote:
When purchasing these phones, make sure they are SIP, as these were
available with several different firmware loads
You may
will be available to you without proper credentials from Cisco
John Novack
Turritopsis Dohrnii Teo En Ming wrote:
Subject: I found an excellent guide: Configure Asterisk VoIP IP PBX SIP Server
with Cisco IP Phones
Good day from Singapore,
Today 18 December 2020 Friday, I found an excellent guide
version to
work with
For learning there isn't any good reason to have the latest of anything
I have a working version of Asterisk 13 with DAHDI and a 4 port T1 card on
CentOS 6, and support a buddy with a TDM 400 or 410 - no issues
YMMV
John Novack
Roy Kidder wrote:
Hello all,
It's been quite
JMO
AstLinux installed on an HP Thin Client is a good choice for someone with
limited knowledge of Linux who wants a less steep learning curve.
YMMV
John Novack
Turritopsis Dohrnii Teo En Ming wrote:
Subject: How to DIY/Setup An Open Source IP PBX Appliance/Server?
Good day from
Hello,
I'm working on converting my 18.0.1 test system from SIP to PJSIP and I've
run into something odd.
I have a queue defined named acme-test that has two agents in it,
PJSIP/7001acme and PJSIP/7002acme.
I have autohints=yes in my acme-intern context, I have not defined hints
for either of
401 should contain an authentication header.
The 401 response should be followed up by a second INVITE containing an
authorization header. Maybe credentials are not setup correctly on the
sip client.
John
--
_
-- Bandwidt
gt;>>>
>>>>>> --
>>>>>> David Cunningham, Voisonics Limited
>>>>>> http://voisonics.com/
>>>>>> USA: +1 213 221 1092
>>>>>> New Zealand: +64 (0)28 2558 3782
>>>>>> --
>>>>>> _
Hello all,
Anyone know an easy way to have the Directory
Application<https://wiki.asterisk.org/wiki/display/AST/Directory+Application>
lookup all the voicemail contexts in the system. Like a global option
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
,u)=WaitTime: 20)
same => n,Set(FILE(${CALLFILE},,,al,u)=Context: alice)
same => n,Set(FILE(${CALLFILE},,,al,u)=Extension: s)
same => n,Set(FILE(${CALLFILE},,,al,u)=Priority: 1)
same => n,Set(FILE(${CALLFILE},,,al,u)=SetVar: John=AWESOME
same => n,Set(FILE(${CALLFILE},,,al,u)=Arc
o
announce_user_count=no
wait_marked=yes
end_marked=yes
music_on_hold_when_empty=no
quiet=yes
;
[xaccel]
type=bridge
record_conference=yes
;
Then calling in I see this
Conference Bridge Name Users Marked Locked Muted
== == == =
xaccel
1,NoOP(Options to $EXTEN)
same => n,Hangup()
I added hints to see if that would make a difference and it hasn't.
I also made a 'Anonymous' peer to see if that would help without any luck.
On Thu, Jul 16, 2020 at 6:11 PM Joel Serrano wrote:
> Hey John,
>
> In one installation I h
I'm implementing a SBC with my Asterisk PBX but the keeps disabling the
trunk group I've configured and I think it may be because Asterisk is
returning a 4r04 to the OPTIONS.
I've created a test context and have put in a wildcard pattern match to try
and catch those options but it doesn't seem to
through
Any help is much appreciated.
Thanks
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806.2604
Cell: 973.390.1090
www.xaccel.net<http://www.xaccel.net/>
CONFIDENTIALITY NOTICE:
This e-mail message, inc
ct that we need some
service -- not necessarily his -- to sign the call before sending it
to our normal carrier, or will the normal carrier -- whoever -- sign
the call if they know the number?
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
Dovid, You could use func_odb + a ODBC Redis driver to keep from having to
shell out.
On Wed, Jul 8, 2020 at 4:37 AM Dovid Bender wrote:
> Hi,
>
> Does anyone know of any projects that would allow you to use Redis in
> place of AstDB? By in place of I don't mean for what Asterisk needs but to
>
ing, is it correct ? ... I tried without the brackets...
that also doesn't work.
If not supported in includes
What is the formatting for timezone in gotoiftime.
GotoIfTime(times,weekdays,mdays,months,[timezone]?[labeliftrue:[labeliffalse]])
Any helps is much appreciated.
Thanks
John Bittner
On 12/06/2020 16:19, Olivier wrote:
It seems a new Linphone 4.2 is to be published next week !
Hopefully, ...
1. its call history is useless to me, it works very poorly with sip
proxying (i.e. asterisk), the design is clunky (no simple list of all calls)
2. it has no simple busy light
On 10/06/2020 15:40, Joshua C. Colp wrote:
You wouldn't be able to access such information from
ast_sip_presence_exten_state_to_str, that function is strictly for
taking in instructions/data and producing the output. The user of it
would need to pass in a value to turn on this new behavior.
fies notifyringing=no and an extension is in a call
then we send out "On a call" instead of "Ringing" so people can see
who is not going to pick the call up.
Author: John Hughes
Last-Update: 2020-06-09
--- asterisk-13.14.1~dfsg.orig/channels/chan_sip.c
+++ asterisk-13.14.1~dfsg/
On Mon, 8 Jun 2020 at 05:18, Markus wrote:
> Hi list!
>
> I'm getting this error frequently:
>
> ERROR[25193][C-0004f387]: cdr_mysql.c:203 mysql_log: Cannot connect to
> database server localhost: (2026) SSL connection error:
> SSL_CTX_set_default_verify_paths failed
>
On 26/05/2020 15:33, Olivier wrote:
Hi John,
1. Could you get any further, in your quest for working BLF with
linphone ?
The patches to get linphone-3.12 BLF working with Asterisk are here:
http://perso.calvaedi.com/~john/linphone-3/
They're pretty damnned trivial:
1. add the "A
Hello,
Anyone know how to set the "To:" in an invite for PJSIP to custom settings. I
got the "from" to be the way I need it.
From:
I have tried a lot of changes to get to this but nothing works.
I am getting this
From: sip:109643...@xaccel.net;tag=42e4a9cb-59af-4d40-a21f-00261afbd3be
To:
Nice, Do you have the code up on GitHub? I'd love to see it.
What's the source of the data? Something API driven I hope?
Have you thought about implementing your project via curl instead of
func_odbc?
On Wed, May 27, 2020, 8:52 PM Saint Michael wrote:
> In a few weeks, no SIP call is going to
Use the ARRAY version of Set.
same = n,ExecIf($["A" = "B"]?Set(ARRAY(C,D)=1,2))
On Tue, Apr 21, 2020 at 3:56 AM Administrator wrote:
> Hello,
>
> we want to use something like
>
> same = n,ExecIf($["A" = "B"]?Set(C=1) & Set(D=2) & ...)
>
> Problem is that result gives C=1) & Set(D=2) & ...
>
>
On 14/05/2020 16:41, Joshua C. Colp wrote:
On Thu, May 14, 2020 at 11:31 AM John Hughes <mailto:j...@calva.com>> wrote:
On 14/05/2020 08:10, John Hughes wrote:
I am having a problem with one of my callers who is using either
g729 or alaw. I can do alaw but not g729 so
On 14/05/2020 08:10, John Hughes wrote:
I am having a problem with one of my callers who is using either g729
or alaw. I can do alaw but not g729 so asterisk should negotiate alaw
right? In fact from the sip debug it looks like it does, but then I
get the dreaded "channel.c:5630 set_f
RTP debugging.
Asterisk 13.14.1 on Debian, using chan_sip.
Hi John,
Maybe a newer version of Asterisk would help? The latest release for
13 is version 13.33. The version you are on was released 3 years ago.
Well, like I said I'm on Debian, using the packaged version. If I want
to upgrade I'll ha
I am having a problem with one of my callers who is using either g729 or
alaw. I can do alaw but not g729 so asterisk should negotiate alaw
right? In fact from the sip debug it looks like it does, but then I get
the dreaded "channel.c:5630 set_format: Unable to find a codec
translation path:
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Your life is like a penny. You're going t
Doug
According to this document, there is no way for me to change the
volume(s) for another user, whereas meetme allows me to do this by
specifying the conference number and user number.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
to stay with meetme. And
I wonder if its a meetme issue?
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- Bandwi
other
participants audio, meetme has this feature which I use frequently.
--
Your life is like a penny. You're going to lose it. The question is:
How do
you spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start
On 23/03/2020 18:51, Joshua C. Colp wrote:
On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote:
Why is asterisk giving an error 500? I can find no reason, there
is nothing in any log.
The sequence number is from the past. The first SUBSCRIBE is
On 23/03/2020 18:51, Joshua C. Colp wrote:
On Mon, Mar 23, 2020 at 2:45 PM John Hughes <mailto:j...@calva.com>> wrote:
Why is asterisk giving an error 500? I can find no reason, there
is nothing in any log.
The sequence number is from the past. The first SUBSCRIBE is
isk", nonce="188b095b", algorithm=MD5,
username="john", uri="sip:jacques@10.27.128.1:5060",
response="bdbc7cbac4453fd643050bf28996a68e"
<----->
--- (14 headers 0 lines) ---
Found peer 'john' for 'john' from 10.27.128.3:5060
<--- Transmitting
On 23/03/2020 11:29, Joshua C. Colp wrote:
On Mon, Mar 23, 2020 at 7:15 AM John Hughes <mailto:j...@calva.com>> wrote:
Hi, in these dark days of COVID-19 lockdown I'm using linphone to
connect to my office asterisk system for working from home.
It's going pr
Hi, in these dark days of COVID-19 lockdown I'm using linphone to
connect to my office asterisk system for working from home.
It's going pretty well but the presence/BLF functions don't appear to work.
In the linphone logs and asterisk debug I find that asterisk is
rejecting linphone's
look for 'mytrunk' as thats the trunk its dialing
On Wed, Mar 18, 2020 at 02:41:51PM -0300, Joshua C. Colp wrote:
> On Wed, Mar 18, 2020 at 2:37 PM John Roman wrote:
>
> > ive enabled logging. aside from a realm error i see on my endpoint, im
> > still not sure whats
;tag=3166828162
To:
;tag=f1b212ab-9b55-4d13-9055-f49ce55f214e
Via: SIP/2.0/TCP
[2605:e000:130a:fb:de1:71fc:e257:6f4e]:60629;branch=z9hG4bKd67297ab9853f9168cf8c8533868ea6b373339;rport
CSeq: 8613 ACK
greetings asterisk users :)
ive just deployed version 17 and migrated as best I can to pjsip. I can
receive calls, and get to my mailbox prompt, however placing calls seems
impossible with the following error on dial:
Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid =
My Apologies Dovid, I think I misunderstood your request.
You don't have the time you need to convert in the format of date string,
Instead you have your users entering via DTMF when they want something to
happen?
On Thu, Feb 13, 2020 at 11:08 AM Dovid Bender wrote:
> John,
>
> Fro
Try using the STRFIME function instead of doing this by hand.
https://wiki.asterisk.org/wiki/display/AST/Function_STRFTIME
*%H*
The hour as a decimal number using a 24-hour clock (range 00 to 23).
*%I*
The hour as a decimal number using a 12-hour clock (range 01 to 12).
On Thu, Feb 13, 2020
to the PRI.
Any ideas would be helpful.
Thanks
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806.2604
Cell: 973.390.1090
www.xaccel.net<http://www.xaccel.net/>
CONFIDENTIALITY NOTICE:
This e-mail message, includi
Ira,
What version of Asterisk are you using, and what channel driver?
There has to be a better way than to create hundreds of peer entries.
On Thu, Dec 12, 2019 at 12:26 PM Ira wrote:
> Hello Jan,
>
> Tuesday, December 3, 2019, 8:49:28 PM, you wrote:
>
> Jan> The next thing to look at is
ing+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
John Runyon
Simply NUC
512-766-0401 x1110
495 Round Rock West Dr, Round Rock, TX 78681
--
https://dev.mysql.com/doc/refman/8.0/en/group-by-functions.html#function_count-distinct
Use something like count(distinct src) instead of count(*)
On Tue, Nov 12, 2019, 07:35 Andre Gronwald
wrote:
> hi,
>
> we want to extract the information when the most callers are entering
> our phone
orum at: https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-use
use Asterisk 16.
>
> On Mon, Oct 7, 2019 at 5:58 AM George Joseph wrote:
>
> On Fri, Oct 4, 2019 at 1:19 PM John Covici wrote:
>
> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10
> system and I am running into the following problem. I need to install
&g
spend it?
John Covici wb2una
cov...@ccs.covici.com
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New
_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-us
On Thu, 1 Aug 2019 at 17:07, Doug Lytle wrote:
> On 8/1/19 5:08 PM, Dovid Bender wrote:
>
> Glenn,
>
> I can't use MySQL as each node currently has MySQL however there is a lot
> of data that is stored locally on each box. I may have to take this route
> if I can't find something else but that
Works for me from Comcast!
John Novack
Doug Lytle wrote:
I'm currently reviewing the Digium wiki on migrating from chan_sip to res_pjip
and I'm trying to access the script that is provided to help with conversion.
https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip
Jerry, What if you specify a higher bitrate to mpg123?
You are limiting it to 8k with the 'r' option.
I convert my source audio files with sox to 16khz signed linear for
wideband hold music.
sox -c1 hold.wav -r 16000 -c 1 -e signed-integer -r 16k hold.raw
Then I rename the .raw file to a
On Fri, 5 Jul 2019 at 14:28, hw wrote:
> I thought about that and checked the configuration I've been using to
> create the certificate, and I can't see anywhere that it would expire
> earlier than after 3650 days. Is there another way to check this?
>
openssl verify -CAfile ca.crt server.crt
Joshua,
Thanks for looking into this, and sorry for not being more detailed.
Running asterisk 16.4.0
I was able to get in touch with an AIphone tech and it turns out that these
issues are known bug on their side.
I will be more detailed next time
Thanks
John Bittner
Xaccel
-Original
r comes up with the menus to put a call on hold. So no audio.
Anyone have any ideas or willing to do some consulting work please let me know
asap.
FYI some captures are attached.
Thanks
John Bittner
CTO
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806
be changing it to give them a 404.
Looks like someone's making a big effort to find provisioning files though.
On Mon, Jun 17, 2019, 13:35 John Kiniston wrote:
>
>
> On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote:
>
>> Anyone know how someone can hack an asterisk box and r
On Sun, Jun 16, 2019 at 3:37 PM John T. Bittner wrote:
> Anyone know how someone can hack an asterisk box and register with every
> single account on the box.
>
> This box only has 3 accounts, with very complex passwords. Have VoIP
> blacklist setup and fail2ban…
>
I've see
is blocked.
Logs do not show any http access, secure or any other fingerprints.
I am going to honeypot this box to see if I can capture there invites.
John
Xaccel
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of Dovid Bender
Sent: Sunday, June 16, 2019 6:59 PM
/sip:ghbhhm@5.79.64.23:9228 d7bf838918 NonQual
nan
Any helps is much appreciated.
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806.2604
Cell: 973.390.1090
www.xaccel.net<http://www.xaccel.
\' from '.*' failed for ':.*'
.* - Failed to authenticate
NOTICE.* .*: Request \'REGISTER\' from '.*' failed for ':.*'
.* - Error to authenticate
NOTICE.* .*: Request \'INVITE\' from '.*' failed for ':.*' .*
John Bittner
Xaccel
From: asterisk-users [mailto:asterisk-users
Too bad.
LOTS of users will still want to continue to use these cards, example the OP!
Good news it probably suppresses prices on used cards!
John Novack
Malcolm Davenport wrote:
Howdy,
That is correct.
The list of supported cards is in the README file (not the -complete package
Doesn't DAHDI 3.0 remove support for a bunch of older cards, including the
TDM400 and 410?
John Novack
Greg Woods wrote:
On Thu, Jun 6, 2019 at 12:17 PM Malcolm Davenport mailto:malco...@sangoma.com>> wrote:
Howdy,
There's a dahdi-linux-complete-3.1.0-rc1+3.1.0-rc1.
for '71.127.239.22:65476'
(callid: 211973110361898-30014604441241@192.168.1.8) - Failed to authenticate
[2019-06-06 15:39:17] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request
'INVITE' from '"as100" ' failed for
'188.214.128.172:5071' (callid: 8e12f1560bfe2c3ed5be895108727c46) - No m
Hopefully, this may help someone in the future.
If I set this before I dial out... it works.
I have always in the past set this on hangup... that does not work anymore.
John
Xaccel
From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
Of John T. Bittner
Sent
appreciated.
Testing on asterisk 16.3.0
John Bittner
CTO
[xaccellogoemail]
380 US Highway 46, Suite 500
Totowa, NJ 07512
Phone: 201.806.2602 x2405
Fax: 201.806.2604
Cell: 973.390.1090
www.xaccel.net<http://www.xaccel.net/>
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