Hi,
In the past when I wanted to back port a patch I would go on to the issue
tracker and find a link to the patches that were uploaded ( I think
through gerrit?). I am trying to see what changes were done for
https://issues-archive.asterisk.org/ASTERISK-26109. It seems the code
changes were
Hi,
We recently had a customer that set up Asterisk with port 5038 open to the
world with standard configs for the AMI (by that I mean they copied and
pasted configs that they saw online). Digging around a bit it seems the
attacker used the AMI action "pjsip show auths" followed by "pjsip show
Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts
at 382 and voicetel if you want an intro.
On Thu, Aug 17, 2023 at 11:50 PM Federico
wrote:
> I am looking for a decent provider of SIP Trunks but it has to pass the
> Stir Shaken token to the next carrier. Does anybody
Josh,
Thanks a lot, that worked!
On Wed, May 17, 2023 at 4:21 AM Joshua C. Colp wrote:
> On Tue, May 16, 2023 at 11:01 PM Dovid Bender wrote:
>
>> Hi,
>>
>> I am trying to use SAY_DTMF_INTERRUPT with Asterisk 20.0.1. I see that I
>> asked about it here https://
Hi,
I am trying to use SAY_DTMF_INTERRUPT with Asterisk 20.0.1. I see that I
asked about it here https://www.spinics.net/lists/asterisk/msg174142.html
and Sean was nice enough to create a patch. I am trying it on 20.0.1 by
doing in the dial plan:
CHANNEL(SAY_DTMF_INTERRUPT)=on
and I get back:
Hi,,
When using a SIP proxy to load balance calls how do you make it that a call
on an attended transfer reaches the same Asterisk box every time? I was
told that in later versions of Asterisk there is some "magic" to make it
work correctly when load balancing.
TIA.
Dovid
--
David,
We had this exact "issue" in the past and were not able to figure out how
to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:
Dial(SIP/1234@1.1.1.1//2.2.2.2)
became:
Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2)
On Kamailio's side in the FORWARD block we added:
# HACK
Hi,
Does anyone have any guides, documents on best practice for "bridging"
multiple Asterisk boxes together so no matter what box a person lands on,
they can be on the same call? I assume the easiest would be to have one box
dial out to all other boxes and bridge them. For example If we have room
Hi,
Does anyone have any tips on how to allow bookmarking of where a caller is
holding using ControlPlayBack? I was trying to use ApplicaitonMap to call
an AGI while using ControlPlayBack but that did not work (as per the docs
some functions/features/applications that won't work with application
David,
Are you getting any RTP from the PSTN for either leg? If not it could be
that they assume you are behind NAT and want to see where the SRC of the
RTP before they send it back. We had a few carriers that did this. The
easiest way to get around it was to play a 0.5 second audio clip to the
Hi,
When using GET DATA in an AGI it seems that the # key ends the input. So if
say I want the user to input 123#456 the system will return 123. I did not
see this in the documentation. Is this a bug, lack of documentation or do I
have a bug in my AGI?
TIA.
Dovid
--
In case anyone else has this issue, the problems was that Asterisk had no
permission to /var/cache/asterisk. As soon as I fixed the permissions there
it worked.
On Tue, Feb 22, 2022 at 2:30 PM Dovid Bender wrote:
> Hi,
>
> I am trying to listen to a file that is stored on a web serv
Hi,
I am trying to listen to a file that is stored on a web server. I did see
Josh's response to a similar question here
https://community.asterisk.org/t/failed-to-create-temporary-storage/80739.I
have used this functionality in the past on other systems wth no issue.
Selinux is disabled and
David,
I vaguely remember having this issue on “newer” versions of Linux. I build
from git and it works every time. I will try to look at my scripts and post
later exactly what I do.
On Mon, Feb 21, 2022 at 20:52 David Cunningham
wrote:
> Hello,
>
> I see some emails about a Dahdi compilation
:
> Does asterisk follow HTTP redirects? If so can you use something like
> tinyurl.com to produce an alternative URL?
>
> Or, base64 encode the URL, and then set a variable with
> Set(url=${BASE64_DECODE(${encodedURL})) ?
>
> Cheers,
> Kingsley.
>
> On Wed, 2022-01-2
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 f
> Today's Topics:
>>
>>1. )
>>2. Re: How to escape the & in BackGround (Doug Lytle)
>>3. Re: How to escape the & in BackGround (Dovid Bender)
>>
>>
>> --
>>
>>
I tried single quotes, double quotes, backslash etc and none of it worked
On Sun, Jan 16, 2022 at 16:11 Doug Lytle wrote:
> On 1/16/22 2:19 PM, Dovid Bender wrote:
>
> Does anyone know a way of telling Asterisk that & is part of the URL and
> to pass it along as a string?
>
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 is a new file and the a=b is not sent
along. I tried doing \& but that did not work. Does anyone know a way of
k
> [Jan 6 08:19:30] -- Executing [s@f2:2] NoOp("Local/s@f1-0002;2",
> "f2 / f1") in new stack
> [Jan 6 08:19:30] -- Executing [s@f2:3] Hangup("Local/s@f1-0002;2",
> "") in new stack
> [Jan 6 08:19:30] == Spawn extension
Steve,
I thought of this but that would mean I would need to add this to the
beginning of every context which I can do, but I was trying to avoid.
On Wed, Jan 5, 2022 at 10:06 PM Steve Edwards
wrote:
> On Wed, 5 Jan 2022, Steve Edwards wrote:
>
> > same = n,
Hi,
I have a hangup handler that's added at the beginning of a call. It logs
all the call details. Using the CONTEXT variable I am always going to get
the context where the code is being ran and not the last context that the
caller is in. Is there any creative way to get the last context at the
Thanks Sean.
On Thu, Dec 23, 2021 at 5:55 PM Sean Bright wrote:
> On 12/23/2021 2:47 PM, Dovid Bender wrote:
>
> Has anyone gotten SAY_DTMF_INTERRUPT to work?
>
>
> Issue created[1] and tentative patch submitted for review[2].
>
> [1] https://issues.asterisk.org/jira/br
Hi,
Has anyone gotten SAY_DTMF_INTERRUPT to work? In 19.0.1 if I do core show
function SAY_DTMF_INTERRUPT it does not show up. As per the documentation
This application will play the sounds that correspond to the given .
Optionally, a may be specified. This will use the language that is
on-Commercial Discussion > us...@lists.digium.com>
> > Subject: Re: [asterisk-users] Exec two commands with ExecIf
> >
> > On Thu, 23 Dec 2021, Dovid Bender wrote:
> >
> > > Is there any way of using ExecIf to run two commands instead of 1? e.g.
> > >
Hi,
I didn't see any documentation for this so I assume it can't be done but I
figured I would check here first. Is there any way of using ExecIf to run
two commands instead of 1? e.g. instead of
Exten 123,1,ExecIf($["FOO" == "BAR"]?BackGround(you-owe))
Exten 123,1,ExecIf($["FOO" ==
Hi,
I am experimenting with arrays in Asterisk. I am looking at
https://wiki.asterisk.org/wiki/display/AST/Function_SHIFT and
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_ARRAY.
So for example I Do Set(FOO(1,2,3,4)=10,20,30,40)
What would be the correct way to get both the
Hi,
I am trying to help a friend that is using Asterisk along with a Python
AGI. In the code I see:
agi.execute(pystrix.agi.core.StreamFile(MY_PATH+'/'+msg))
try:
agi.execute(pystrix.agi.core.RecordFile(rec_file,
format='wav',
Sorry about that. Too early to be working on a Sunday morning.
On Sun, Oct 10, 2021 at 10:01 AM Doug Lytle wrote:
> On 10/10/21 9:31 AM, Dovid Bender wrote:
> > Hi,
> >
> > I see that you have pricing for the 12 C1000-48T-4X-L C
> >
> >
>
> I take it t
Hi,
I see that you have pricing for the 12 C1000-48T-4X-L Catalyst 1000 48port
GE, 4x10G. How much is the smartnet 8/5 support for them. Also what would
be the lead time if we purchased them in the next week?
TIA.
Dovid
--
_
Hi,
If I have in extensions.conf includes to files that does not exist Asterisk
stop loading all other files. Say for instance I have:
#include one.conf ; Exists
#include two.conf ; Does not exist
#include three.conf ; Exists
If two.conf does not exist, even if three.conf exists, asterisk will
Hi,
I found
https://markmail.org/message/xh5sbqvsgwywrjje#query:+page:1+mid:vbzl4hup6jawrzup+state:results
and it seems to have answered my question. By adding an expires value to
the response Asterisk seems to cache it for the time specified.
On Sun, Sep 5, 2021 at 6:23 AM Dovid Bender wrote
Hi,
Is there any way to set the default expiration for the media cache? After
looking at the sqlite3 db it seems that asterisk by default sets the
expiration to the time that the file was accessed so the file is never
cached locally and everytime the file is played, it's downloaded again.
--
On Tue, Aug 10, 2021 at 8:19 AM wrote:
> On 8/10/2021 6:06 AM, Antony Stone wrote:
> > On Tuesday 10 August 2021 at 12:40:39, Dovid Bender wrote:
> >
> >> Hi,
> >>
> >> Is there any way in Asterisk to say an ordinal number?
> > My so
Hi,
Is there any way in Asterisk to say an ordinal number? For instance if I
were to pass the application 1 it would say first. If I were to pass it 2
it would say Second etc.? I know that Voicemail has this when hearing your
message count but I have not seen any application that allows this from
the prerotate
> > script of logrotate (in case you use any of these in your env).
> > Certainly this is not a final solution but it is already something that
> > doesn't depend on an asterisk patch.....
> >
> > On Thu, Jul 8, 2021 at 3:58 PM Dovid Bender wrote:
> >
>
On Fri, Jul 9, 2021 at 10:49 AM George Joseph wrote:
>
>
> On Thu, Jul 8, 2021 at 3:58 PM Dovid Bender wrote:
>
>> Hi,
>>
>> We have a project where people will be making payments over the phone. I
>> would like block Asterisk from logging any time th
Hi,
We have a project where people will be making payments over the phone. I
would like block Asterisk from logging any time the system is processing a
card. So be it SayDigits(123456789), when the user enters DTMF or when I
pass a card number as a variable to an AGI etc. I assume this affects
Sean,
Yes that works. It's an "ugly hack". Would this be classified as a bug or
feature?
On Wed, Jun 30, 2021 at 9:30 AM Sean Bright wrote:
> On 6/30/2021 8:55 AM, Dovid Bender wrote:
> > [2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779
> > ast_o
Hi,
I am trying to use ControlPlayBack but pass along some values in the GET
request. For instance if I try
ControlPlayBack(http://localhost/test.gsm?foo=bar)
I get an error in Asterisk
[2021-06-30 08:46:43] WARNING[9661][C-000c8eaa]: file.c:779
ast_openstream_full: File
Hi,
It's been a very long time since I dealt with a along lines. Does anyone
know if there is a way to "pass though" a hook flash? I am working on a
project where there will be one FXS and one FXO. I want if there is call
waiting for the phone connected to the FXS to be able to hit the hook and
t;
> I can live with the "junk" warnings and the quiet playback, but it cannot
> be crashing on valid id3 content (and I have no control over the remotely
> hosting files).
>
> Thanks for any pointers in the right direction!
>
> On Wed, 5 May 2021 at 22:11, wrote:
>
>>
Hi,
Is anyone aware of any way of getting ControlPlayBack to work with an
amazon S3 bucket? I know I can put nginx in the middle but I am trying to
avoid that.
TIA.
Dovid
--
_
-- Bandwidth and Colocation Provided by
25 Feb 2021, Dovid Bender wrote:
>
> > Other than creating an AGI that opens a file to get a json object to set
> > as variables is there any other easy way to set variables for a call
> > when it starts?
>
> Regardless of if there is a way in dialplan, I'd vote for an
Other than creating an AGI that opens a file to get a json object to set as
variables is there any other easy way to set variables for a call when it
starts?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
H All,
We have a carrier that sometimes when sending DTMF will go backwards with
the timestamp. If we look at the Sequence numbers and timestamp the
sequence numbers are in order however the timestamps are not. For instance
seq: 28867 timestamp: 593496832 DTMF 1
seq: 28868 timestamp: 593496832
Check to see if you are being sent INVITES from random IP's. If the
attackers don't have their scripts set up correctly they don't respond
correctly to your responses which Asterisk does not like.
On Wed, Jan 13, 2021 at 9:38 AM Fourhundred Thecat <400the...@gmx.ch> wrote:
> Hello,
>
> I am
Shaun,
Thank you. It seems like it is:
Span 1: WCT13x/0 "Wildcard TE131/TE133 Card 0" (MASTER) ESF/B8ZS RED
ClockSource
On Tue, Jan 12, 2021 at 6:32 AM Shaun Ruffell wrote:
> On Tue, Jan 12, 2021 at 05:31:31AM -0500, Dovid Bender wrote:
> > Yes but it does not tell me if
Yes but it does not tell me if the actual hardware is being used or if its
using the kernel (in place of hardware).
On Wed, Jan 6, 2021 at 4:50 PM Steve Edwards
wrote:
> On Wed, 6 Jan 2021, Dovid Bender wrote:
>
> > The question is if it's using the card or the card or
>
>
> pbx10:newline:13:25:02> timing test
> Attempting to test a timer with 50 ticks per second.
> Using the 'timerfd' timing module for this test.
> It has been 1000 milliseconds, and we got 50 timer ticks
>
>
I get:
as2-c3-njr2*CLI> timing test
Attempting to test a timer with 50 ticks per
Hi,
I have a box that I suspect had timing issues. I added a TE131 to see if
that would help. Is there any way for me to verify that Dahdi is using the
card for timing and not the kernel?
--
_
-- Bandwidth and Colocation
.st...@asterisk.open.source.it> wrote:
> On Tuesday 29 December 2020 at 17:12:45, Dovid Bender wrote:
>
> > I then watched the PID of asterisk for CPU usage and every so often (3-5
> > seconds) the CPU usage would go from 0.0 to 5.X. I know this is not a lot
> > as it's just
All,
As I have written in the past I am chasing down a CPU issue with Asterisk.I
took a box running 16.13.0 and unloaded all possible modules. The only ones
I was not able to unload were:
pbx_spool.so
res_adsi.so
res_odbc_transaction.so
res_sorcery_memory_cache.so res_speech.so
s and the quiet playback, but it cannot
> be crashing on valid id3 content (and I have no control over the remotely
> hosting files).
>
> Thanks for any pointers in the right direction!
>
> On Tue, 3 Sept 2019 at 16:20, Dovid Bender wrote:
>
>> Ludovic,
>>
>>
>>
>> How did you want to handle the backend of the phone system? All database
>> driven and youd code up the IVR? Or do you want it all handled with
>> something like the ARI?
>>
>> Dan
>>
>>
>>
>> On Tue, Dec 1, 2020 at 7:12 PM Dovid Bender
Doug,
That just has info on the software side. I was wondering what kind of
hardware I can get in place of using the CPU for timing.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new
For older versions of Astrerisk (1.8, 11, 13) what is everyones
experience when it comes to core count vs higher speed cpu's? We seem to be
hitting some cpu issues. In the past we went for more cores but now I
wonder if we should go with fewer cores and higher clocked CPU's.
--
Hi,
We are having some issues and we are trying to rule out any timing issues.
What is the recommended hardware for Dahdi 2.X and Dahdi 3.X?
TIA.
Dovid
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
nes you suggest could be the best way forward.
>
> Thank you for that.
>
>
> On Thu, 3 Dec 2020 at 13:01, Dovid Bender wrote:
>
>> David,
>>
>> Does Asterisk send a 180 or a 183 with SDP? We have found that using
>> these two lines help (where xc is a 500ms
David,
Does Asterisk send a 180 or a 183 with SDP? We have found that using these
two lines help (where xc is a 500ms blank sound file)
Exten => _X.,n, Progress()
Exten => _X.,n, Playback(xc,noanswer)
On Wed, Dec 2, 2020 at 4:30 PM David Cunningham
wrote:
> Hello,
>
> We have a problem with
Hi,
Is anyone aware of any way of changing the contact header on a call? We are
sending 911 calls to a provider and they require that the contact be the
call back number. I tried:
Set(PJSIP_HEADER(update,contact)=)
But the came back with:
No headers had been previously added to this session.
If
nd is strict about accepting RTP from
> the specified source and won't accept it. Have you any suggestions to solve
> that problem?
>
> Thank you.
>
>
> On Fri, 30 Oct 2020 at 14:49, Dovid Bender wrote:
>
>> Why not use OpenSips/Kamailoo in between? Where you want 1.1
Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio
On Thu, Oct 29, 2020 at 20:44 David Cunningham
wrote:
> Hello,
>
> Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
>
Anyone have any other ideas?
On Tue, Oct 27, 2020 at 1:27 PM Dovid Bender wrote:
> Jon,
>
> We are only using FastAgi. On the second system (running Asterisk 16)
> there are no agi's running (just some bash scripts on call hangup). I did
> add some hackey code (netstat -nua
27 Oct 2020 12:52:47 -0400
> Dovid Bender escribió:
>
> > Hi,
> >
> > Sorry in advance that I am emailing the users list and not the biz list I
> > think I will find my target audience here. We are looking to hire a
> > consultant to help us figure
Hi,
Sorry in advance that I am emailing the users list and not the biz list I
think I will find my target audience here. We are looking to hire a
consultant to help us figure out an issue. We are having what seems are
"random load" issues with bare metal boxes that are dedicated to Asterisk
and a
All,
I am stuck with a specific install using chan_sip and Asterisk 11.25.3. We
have nat=no which from what I understand means that Asterisk will go by
whatever it see's in the SDP and not look at the source IP+port from where
the traffic is coming from. We have a call flow where we send a
Asterisk will try calling both at once. As soon as one is answered it
cancels the call to the other. What you can do is for extension 101 to put
it in it's own context and then call the agi from the h extension. So
something like this:
[from-internal]
exten = 514316,1,Answer()
same =>
Is there anything in the Asterisk logs? Which side sends the BYE? Were you
able to capture the traffic with sngrep/wireshark to see if any side
stopped sending/getting RTP? What did the other side see?
On Mon, Sep 21, 2020 at 3:22 PM Roberto <
roberto.med...@gasparimsantos.com.br> wrote:
>
Hi All,
I built a system which allows people to call a phone number and listen to
various online media streams (train yards, radio stations etc). I use
ffmpeg + MusicOnHold to play the streams. The system also allows callers to
hear pre recorded content. Normally about 250 calls equates to about
Hi,
We have a client that is looking for a CPCI or ATR system. If someone from
sales could get back to me that would be greatly appreciated. I can be
reached at 848-210-0001 or 914-600-2000.
Thanks in advance.
Regards,
Dovid
--
; Are you looking for a general caching solution or you specifically need to
> use Redis? At Thirdlane we use Memcached which works just fine for our
> purposes.
>
> Regards, Volodya Ivanets
>
> ----------
> *From: *"Dovid Bender"
> *To: *&q
years. Is there any reason why there was never a
push to have this added to Asterisk? It seems like a no brainer and we
would pay for development of it. I am asking before I start in case there
is a reason why it was not done.
On Wed, Jul 8, 2020 at 7:36 AM Dovid Bender wrote:
> Hi,
>
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 store
values. For instance for CNAM currently we need to use an AGI to connect to
redis to pull CNAM. So in place of:
On Mon, Jun 29, 2020 at 6:46 AM Joshua C. Colp wrote:
> On Sun, Jun 28, 2020 at 2:26 PM Dovid Bender wrote:
>
>> Hi,
>>
>> We have a box up and we are starting to see a lot of "Exceptionally long
>> queue length queuing" in the logs. From all the res
Hi,
We have a box up and we are starting to see a lot of "Exceptionally long
queue length queuing" in the logs. From all the research so far it seems
like this leads to their systems crashing and being unreachable. In our
case the box remains up and takes calls. We are running Asterisk 16.6.1. We
What is the application that you are missing?
On Sun, May 17, 2020 at 01:32 Saint Michael wrote:
> I want to see the help when I type core show application , and it's
> not available. This is asterisk 16 from sources. I have libxml2-dev
> installed. Ubuntu 19
> What am I missing?
> Philip
>
Hi All,
I vaguely remember someone at Astricon making the case for having multiple
containers/vps each running asterisk vs using asterisk direct on bare
metal. Something about getting better performance. Does anyone have any
insight on this?
TIA and stay safe
Dovid
PS I know vps != containers
n 4/26/20 10:48 AM, Dovid Bender wrote:
> > Hi,
> >
> > Looking at
> > https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there
>
> > is an option for admin_toggle_mute_participants however the non admin
> > users can still toggle togg
Paddy,
Why not use local extensions? You can do something like this.
Exten => s,1,Dial(Local/set1@call_all/set2@call_all
/set3@call_all)
[call_all]
Exten => set1,1,Dial(SIP/100/101/102/103/104/105
Exten => set1,1,Dial(SIP/106/107/108/109/110/111
Exten => set1,1,Dial(SIP/112/113/114/1015/116/117
Joran,
A "hack" would be to issue the command with one thread and with the other
tail the log and lookout for "WARNING".
On Mon, Apr 27, 2020 at 9:24 AM Jöran Vinzens wrote:
> Hi All,
>
> I hope someone can give me a hint.
>
> We try to reload the asterisk dialplan config using ansible
Hi,
1) Is there any reason why max_pseudo_channels defaults to 512? I want to
increase it by default but at the same time don't want to outsmart the
developers if they had a good reason for it.
2) I had a look at
http://lists.digium.com/pipermail/asterisk-users/2014-March/282607.html but
that did
Hi,
Looking at
https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration there
is an option for admin_toggle_mute_participants however the non admin users
can still toggle toggle_mute. Is there any option for the admin to disallow
non admins from using toggle_mute to unmute themselves?
to smooth out CPU usage on one of my servers.
>
>
> On 4/22/20 2:01 PM, Dovid Bender wrote:
> > Hi,
> >
> > I have an Asterisk box which has an IVR that plays random gsm files. The
> > box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk
> > CP
:21 PM Telium Technical Support
wrote:
> Could some calls be arriving with a different codec? (Is transcoding
> causing the spikes)? Are you limiting codecs to match your audio files?
>
>
>
> *From:* asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Be
Hi,
I have an Asterisk box which has an IVR that plays random gsm files. The
box has SSD's and two CPU E5-2695 v2 cpus with 64GB ram. The Asterisk CPU
usage along with the load seems to jump around. With about 500 callers it
hovers between 250-400% CPU (so 2.5 to 4 cores) which seems reasonable.
Are you using NFS? Any ODBC connections?
On Tue, Apr 21, 2020 at 10:23 AM Stefan Viljoen
wrote:
> Hi all
>
> I'm running an Asterisk on an Intel XEON E5-2660 virtual with Centos 7 -
> 32GB RAM.
> When I approach about 320 channels, I -sometimes- get thousands of these
> messages suddenly
On Mon, Apr 13, 2020 at 10:45 AM Joshua C. Colp wrote:
> On Mon, Apr 13, 2020 at 11:38 AM Dovid Bender wrote:
>
>> Josh,
>>
>> What should Asterisk do if one of the real time methods fail? I have in
>> extconfig.conf
>> musiconhold => curl,http://localho
idered a failure and it should then move
over to the next rt engine?
On Thu, Jan 2, 2020 at 7:06 AM Joshua C. Colp wrote:
> On Sun, Dec 29, 2019 at 10:22 AM Dovid Bender wrote:
>
>> Hi,
>>
>> Is there any way in Asterisk to have multiple forms of real time for the
>&g
Hi All,
Does anyone have stats on how many callers they can get on a single
ConfBridge per core? I was testing on Digital Ocean with 30 users in a room
and I was shocked by how well it kept up with 40 callers. I am trying to
figure out how scalable it is on VPS's vs using bare metal.
TIA.
Hi,
I am using the ICES application and one issue we are having is the carrier
is timing out because we are not sending RTP. I did try RTP keepalive and
that did not help. Is anyone aware of a way to have Asterisk send a fake
RTP packet (as in a real RTP packet with no audio) in place of RTCP
Hi,
We have a requirement to build a cluster that can handle 30k calls. The
system is going to play one of 15,000 sound files. In the past we had no
issue with Asterisk doing a few hundred calls. When we went above that
Asterisk melted (this is going back quite a few years). FreeSwitch ended up
Hi,
It's been forever since I dealt with POTS lines. We have a client that
needs FXS and FXO support. If memory serves correct we used the TDM400P
with fxs_gs/fxo_gs. What's the equivalent of that card today?
TIA.
Dovid
--
_
))
On Thu, Feb 13, 2020 at 2:12 PM Dovid Bender wrote:
> John,
>
> That is correct. I am trying to figure out why Asterisk is executing the
> set part of the execif, if it's coming back as false.
>
>
>
> On Thu, Feb 13, 2020 at 2:10 PM John Kiniston
> wrote:
>
>
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,
>>
>> From looking at the wiki won't STRFIME just give me what I
sk-users [mailto:asterisk-users-boun...@lists.digium.com] *On
> Behalf Of *Dovid Bender
> *Sent:* Thursday, February 13, 2020 4:47 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Help with FUNC_MATH
>
>
>
> Hi,
>
>
>
> I hav
mber using a 12-hour clock (range 01 to 12).
>
> On Thu, Feb 13, 2020 at 3:49 AM Dovid Bender wrote:
>
>> Hi,
>>
>> I have some dialplan code that is trying to convert 12 hour time with
>> AM/PM to 24 hour format. The code has something like this:
>> E
Hi,
I have some dialplan code that is trying to convert 12 hour time with AM/PM
to 24 hour format. The code has something like this:
Exten =>
2,1,ExecIf(${MATH(${HOUR_SELECTED}<12)}?Set(HOUR_SELECTED=${MATH(${HOUR_SELECTED}+12,int)}))
Earlier on in the dialplan HOUR_SELECTED is set to 12. When
Hi,
Is there any way in Asterisk to have multiple forms of real time for the
same object? For instance my main source for real time is MySQL. I want a
fail over that if a mailbox is say not in the MySQL database for Asterisk
to try via curl.
TIA.
Dovid
--
So long as the tcp socket is open your SBC should send the call back over
the same socket. Now it can be that your SBC is seeing the socket as
timing out. If you are using Kamailio you can have it send tcp keep alives
every so often so that the socket stays up.
On Fri, Dec 27, 2019 at 10:41 AM
1 - 100 of 524 matches
Mail list logo
