It worked with my test. I'm on Asterisk 18.19.0
-- Executing [517xxx@voipms:4] System("IAX2/voipms-15815", "asterisk
-rx 'core restart now'") in new stack
-- Remote UNIX connection
Asterisk uncleanly ending (0).
Executing last minute cleanups
== Destroying musiconhold processes
>> Is there a dial plan call that can "exit asterisk" or completely reload
>> everything - killall active calls and start over ?
Using system() you could issue a asterisk -rx 'core restart now'
Doug
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>>> How do we get this working
For the time being, go back to 18.14.0
Doug
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In the old days when I was using console/dsp, I would have to use
alsamix from the console to verify that the output wasn't muted. Maybe
it's still the same.
Doug
On 9/7/23 03:43 PM, Jerry Geis wrote:
ok switching to "Console/default" does show the text
--- <("<) --- Call to device
On 9/6/23 03:23 PM, Jerry Geis wrote:
I am trying to just play on PulseAudio actually.
This used to work - I have just recently updated to 18.18.0, so I'm
puzzled.
All of my Asterisk installs are running in virtual machines, so I have
no way to test.
Doug
--
What is the device that you're connecting to?
Doug
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In a past work life, I did use console/dsp to connect to a sound card that
hooked up to a bogan paging amp. I still have access to the programming and
everything I have show as using a lower case c for console
Doug
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>>> hi Doug - so what device do you use? I am getting and error for Console/dsp
I don't use it; just figured I'd try to help.
Doug
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>>> Thanks doug - I did that - still showing XXX for chan_console
Just to verify that you did rerun configure after installing the libraries?
Doug
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On my debian 11 install I needed to install
portaudio19-dev
Doug
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>>> I am creating a dialplan where a single user (Alice) has two offices. Both
>>> of her phones should ring if her extension is called.
On my home Asterisk, I have created a home queue and made both of my phones a
member. The first phone that picks up get that call.
Doug
--
On 6/17/23 08:47, Steve Matzura wrote:
Both Background() and WaitExten() allow the caller to enter DTMF
digits. Asterisk then attempts to find an extension in the current
context that matches the digits that the caller entered. If Asterisk
finds a match, it will send the call to that
On 6/16/23 20:29, Steve Matzura wrote:
As always, thanks in advance for a kick in the right direction.
For both capabilities, you can use Background() instead of Playback()
for audio prompts. Background() allows for interrupting the prompts and
continue on with your dialplan.
Doug--
On 5/28/23 14:20, Steve Matzura wrote:
Who controls how many times an incoming call from an external (DID)
provider will ring before Asterisk picks up the call and handles it
internally
Asterisk and this is defined with your timeout on the dial command, mine
is 26 seconds so around 5 rings.
On 5/26/23 01:15, Fourhundred Thecat wrote:
how do I fix this?
What do I have to do to "register" denoise ?
confbridge.conf states:
"Requires func_speex to be built and installed."
I am guessing you have not fulfilled that requirement.
Doug--
On 5/24/23 09:56, Steve Matzura wrote:
I don't understand your explanation because in the two files whose
contents I posted, there's nothing routed to anything called just 's'.
However, I've seen that in the error messages and it stumped me, too.
No 'start' either.
Steve,
Please make sure
On 5/24/23 08:03, Steve Matzura wrote:
*** extensions.conf ***
[general]
[globals]
; Make sure to include inbound prior to outbound because the
_NXXNXX handler will match the incoming call and create a loop
include => voipms-inbound
include => voipms-outbound
[voipms-outbound]
On 5/23/23 19:22, Steve Matzura wrote:
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected
because extension not found in context 'voipms-inbound
Steve,
Could we see your dialplan for voipms-inbound?
I'm using voip.ms as well, but have not converted from chan_sip yet. My
On 4/6/23 01:34, Fourhundred Thecat wrote:
my question is, how can I log this filename in my cdr ?
Set(CDR(userfield)=yourcontent)
Doug
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>>> there are new versions of Asterisk but mailing list is empty
I think they've been having issues, I've noted recent mail coming across that
was from several days ago.
Doug
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On 11/27/22 09:22, Greg Troxel wrote:
Thanks for posting. As I'm running asterisk on a PC Engines apu2, I
don't need the details as it is obviously unworkable, but it's great to
see non-cloud progress.
Greg,
Just a note,
This would work if you have the API server running on a Linux x86 box.
Everybody,
I've recently discovered openai/whisper and have been trying in earnest
to get this working with Asterisk for voicemail transcriptions
(Currently using the NerdVittles script with IBM Watson)
https://github.com/openai/whisper
After spending several hours today, I've successfully
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?
Try enclosing the URL in single quotes,
Doug
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>>> asterisk doesn't support .ogg file format (digged through
Yes it does, if it's complled in with it.
Under make menuselect
=> Format Interpreters
You'll see the development libraries that need to be installed before
re-compiling for ogg playback support
Doug
--
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output-ogg)
If the actual filename is output.ogg then the code should be
exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output)
You'll also need to confirm that you compiled Asterisk with Vorbis support.
Doug
--
>>> exten => _[*+#0-9].,n,Playback(/var/lib/asterisk/sounds/output.ogg,)
Do not use the .ogg when describing the filename.
Doug
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>>> but if the called hangs up prior the timeout for the voicemail, the
>>> Subrouting "noanswer" will not called...
You can use the h priority for that.
https://wiki.asterisk.org/wiki/display/AST/Special+Dialplan+Extensions
Doug
--
>>> so I re-did make and make install and then a full asterisk restart, but
>>> I still got the same "missing dependency: res_fax" error in the log.
You should probably do a
make distclean
And then run configure again before re-compiling.
Doug
--
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 this is an ps moment *grin*
Doug
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>>> How do I do that ? I want all 3 ringing at the same time - and then as they
>>> answer they are brought into the conference.
I'd use call files,
Others I'm sure would use AMI.
Doug
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According to the wiki, you can disable the timestamp
record_file_timestamp
Append the start time to the record_file name so that it is unique.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Configuration_app_confbridge
Doug
--
>>> I do not want to build a SIP server / PBX myself which can itself perform
>>> call hold
>>> & transfer etc (I know how to do that with Asterisk)
I assumed we were talking about an Asterisk server.
Ignore what I just suggested,
Doug
--
>>> I'm looking for something which I can place in the network path between the
>>> client and the server, which can send these call control commands on to the
>>> server, so that it can then put calls on hold, transfer them, etc.
Install Flash Operator Panel
https://www.fop2.com/
Doug
--
>>> Asterisk Project Security Advisory - AST-2021-008
Downloading asterisk-16-current.tar.gz is still showing Asterisk 16.19.0
Doug
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Mike,
The below link turned up for me in a Google Search
https://www.voip-info.org/asterisk-config-chandahdiconf/
Doug
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On 3/7/21 1:43 PM, Greg Troxel wrote:
So I wonder if your asterisk instance is connecting to the PSTN as a
top-level carrier, or, more likely, I am confused in some way.
Greg,
I think this is the case for quite alot of those here.
For me though, I just manage the on premise PBX and my
>>> OK, both combination worked but still silence until the all numbers are
>>> dialed.
I have never used the U option on the dial command to call a sub-routine,
Doug
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>>> Gosub(check-number-forwarding,Dial(SIP/718xxpstn-5665,20,U(atb-sub))
>>> Is ARG1 = atb-sub ?
No.
My complete line
exten => _45XX,1,Set(_ARG1=${EXTEN}
same => n,Gosub(check-number-forwarding,s,1(${ARG1}))
So, if someone were to dial a 4 digit number starting with 45 (i.e.
>>> How do you enable the phone speaker on the Gosub?
>>> I had:
>>> Dial(SIP/718x@pstn-5665,20,m(default)M(atb))
You can provide variables to your gosub routine, for an example
Gosub(check-number-forwarding,s,1(${ARG1}))
Doug
--
>>> app.c:280 ast_app_exec_macro: Cannot run 'Macro(atb)'. The application is
>>> not available.
Macros are no longer built by default in Asterisk 16. This was documented in
the UPGRADE.txt file
app_macro:
- The app_macro module is now deprecated and by default it is no longer
built.
Review your features.conf file in /etc/asterisk
Doug
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>>> Also, you will need a TFTP server working on your Asterisk box
My suggestion would be to get a refurbished Polycom VVX 301 phone (With power
brick if no POE is avaiable) for around $27 US.
Doug
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The wiki page has some information on timing and troubleshooting
https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces
Doug
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On 12/9/20 2:03 AM, Dmitry Melekhov wrote:
But because Centos is declared dead, what is best choice ? Oracle?
Ubuntu?
And for those that have no idea as to what he is referring to (I
didn't), here is the Register article
https://www.theregister.com/2020/12/09/centos_red_hat/
Doug
--
On 8/8/20 8:35 AM, Jerry Geis wrote:
The VM is Intel box (host) and the physical box is a celeron. So
something is not right there.
What would be a good ./configure option that asterisk can compile with
on the VM image so this illegal instruction does on occur ?
Jerry,
Under Compiler Flags
On 6/26/20 4:16 PM, Antony Stone wrote:
Where can I set this threshold?
/etc/asterisk/logger.conf
; All log messages go to a queue serviced by a single thread
; which does all the IO. This setting controls how big that
; queue can get (and therefore how much memory is allocated)
; before new
>>> Is there any way I can tell Asterisk that an ODBC connection problem is a
>>> fatal error
Your be best bet would be to do that check in the script that starts up
Asterisk and maybe a CRON job that periodically tests connectivity.
Doug
--
>>> other than using the System() command?
Not that I am aware of,
Doug
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On 6/19/20 4:23 AM, basti wrote:
Fax is not send. No Sip stuff is show in log.
I don't know what is wrong here.
Best regards
Basti,
This really belongs on the iaxmodem mailing list or the HylaFAX+ Mailing
list. Lee Howard is the author of both packages and very responsive.
Doug
--
>>> Instead, the call still terminates if mysql cannot be reached.
I just tested this, I'm using cdr_odbc, by shutting down mysql and I did not
experience the call being dropped.
The console logged the mysql failure, but the call continued.
You may want to consider moving to cdr_odbc instead.
On 6/5/20 12:24 PM, Marek Greško wrote:
How can this behavior been overriden? I do not expect this is problem
on provider side, since it was working normally using chan_sip.
Console output and dial plan snippets are always useful when diagnosing,
Doug
--
On 5/29/20 2:28 AM, Administrator wrote:
You could also use DEVICE_STATE
I am using DEVICE_STATE to identify when a phone is in use:
exten => s,n,GosubIf($["${DEVICE_STATE(SIP/${ARG1})}" =
"INUSE"]?SHOWBUSY,s,1(${ARG1}))
I'm trying to figure out the best way to display that information to
>>> But if you've already got the caller on the phone, then you might consider
>>> the CONNECTEDLINE function in Asterisk...
And that we don't.
It's the third party that would like the notification the the destination phone
is currently busy with another call. CONNECTEDLINE only functions
Everybody,
I've had a request from my manager that I figure out how to get our Asterisk
13.x system using chan_sip to be able to display on the Polycom VVX series
phone display (firmware 5.9.5), when an extension is called and the person on
the other end is on the phone.
He said, "Our old
On 5/25/20 5:56 AM, Mitul Limbani wrote:
Maybe you can have it uploaded on GitHub.com as a repository ?
With a README.md file on how to install it for PHP7 ?
Anybody that would like to do this would be most welcome.
I have no plans on supporting it.
Basic instructions and attachment will
Everybody,
I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a
dozen years, it was easy to configure and didn't requite installing
'connectors' on anything or adding tables on the DB server.
It's based off of PHP5 and the only reason I still keep around a Debian
7
On 5/16/20 9:57 AM, Michael Maier wrote:
On 15.05.20 at 14:31 Doug Lytle wrote:
Google says Round Trip Time
https://www.voip-info.org/asterisk-rtcp/
That doesn't answer my question (I know the abbreviation RTT). Therefore I'm
trying again:
I'm just wondering what the RTT *exactly* means
Google says Round Trip Time
https://www.voip-info.org/asterisk-rtcp/
Doug
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On 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 toggle_mute. Is there any option for the admin
to disallow non
>>> All the calls are using ulaw. The files that I am playing are gsm. I
>>> suppose doing a file convert with sox to .ulaw may help but it should be
>>> able to do 500 calls without an issue. Can it possibly be a bug? if not how
>>> do >>> I profile which call(s) can be causing the spike?
>>> Can I adjust the talk or listen volume for another user?
I've never used the volume controls, but it would appear.
https://wiki.asterisk.org/wiki/display/AST/ConfBridge+Configuration
Doug
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>>> I never moved to confbridge because they don't have an option for
>>> controlling the volume of other
>>> participants audio
I have menu options in my confbridge configs that has increase and decrease
conference volume.
I'd still configure a small confbridge and test if you still have the
>>> he problem is that there is some sort of distortion in the audio
Has been been going on for a while or is this a new setup? Do you have a
timing source?
I bit the bullet around a year ago and moved to CONFBRIDGE; it wasn't as
horrible as I thought it would be to setup.
Doug
--
>>> How do I do that?
If you are using your package manager to install Asterisk & Dahdi, then I would
not suggest that you compile.
Doug
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>>> I saw something about needing to SIGN the dahdi modules. How do I do that ?
>>> If that is the solution.
Just a guess,
Recompile Dadhi.
Doug
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My Asterisk 13 IAX2 trunk posted below:
type=friend
trunk=yes
allowcallerid=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=my.super.duper.host
username=my.super.duper.username
secret=my.super.duper.secret
context=sip
qualify=500
qualifysmoothing=yes
requirecalltoken=no
trunk=yes
>>> I am trying to troubleshoot two Asterisk servers that have an IAX2
>>> trunk between them.
Carlos,
Had caller-id ever worked between these two systems?
Doug
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>>> Reviewboard is a legacy site and will likely be shutdown. Is there a reason
>>> you are wanting to visit it?
After seeing Olivier's post about his recent failures on compile and it
referencing NBS (Network Broadcast Sound), which I had never heard of, I was
googling to find out more and
Under Firefox, browsing to https://reviewboard.asterisk.org I get
Warning: Potential Security Risk Ahead
Firefox detected a potential security threat and did not continue to
reviewboard.asterisk.org. If you visit this site, attackers could try to steal
information like your passwords, emails,
I understood that part, I was hoping to understand why.
In the past, I've used the PSTN lines to connect two Asterisk systems for
extension to extension calls and was able to route source and destination
extensions via the dial-plan, just by parsing the assigned CID.
Was thinking that may be
>>> I desire to make a call from my system looking like it comes from 4452 and
>>> call the outside number
If you have control over your CID with your provider, you can use
Set(CALLERID(number)=4452)
Otherwise, you cannot.
If you would provide us with what you are trying to accomplish, maybe
>>> I can make calls over a SIP trunk as SIP//number
>>> I am trying to make calls over an extension thought using the same format
>>> SIP/4452/number - its not working
No,
Extension to extension calls would be:
Dial(SIP/${EXTEN])
My extension to extension dial line is
exten =>
>>> Is there some control character(s) for the CLI to interpret everything in
>>> between as a single argument?
I think you can typically use tab completion when working with spaces or you
can escape the space with a back slash
For example Doug Lytle would be
On 12/24/19 10:34 AM, Sean Bright wrote:
On 12/24/2019 9:02 AM, Doug Lytle wrote:
[Dec 24 07:48:46] WARNING[10679] res_calendar_caldav.c: Unknown
response to CalDAV calendar calendar.name.here, request REPORT to
/dav/username/Calendar: Server certificate changed: connection
intercepted
Everybody,
For a while now, I've had a small home Asterisk setup to connect to my
Zimbra mail server's calendar. Making an entry on the calendar would
cause Asterisk to schedule a wakeup call at the time of the calendar entry.
The Zimbra mail server uses LetsEncrypt for the SSL Certs and
On 12/13/19 11:48 AM, Julian Beach wrote:
Hello Doug,
Friday, December 13, 2019, 11:03:37 AM, you wrote:
This is exactly what I do - “press 1 for a human”
Works great
I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call
On 12/12/19 6:55 PM, Adam Goldberg wrote:
This is exactly what I do - “press 1 for a human”
Works great
I do this as well, but I also do a database lookup to see if the number
is on our speeddial list and if so, pass the call directly on without
the IVR prompts.
Doug
--
On 11/26/19 12:31 AM, Jonathan H wrote:
Yes, I know I post similar back in January, but there was no response
back then and I was hoping things might have changed :)
I'm using IBM's Watson for voicemail transcriptions, they allow 500
minutes per month for speech to text on the Free/Lite plan.
On 10/30/19 12:10 AM, Fourhundred Thecat wrote:
Does asterisk not have some internal function to send email ?
It does so for voicemail.
Is there perhaps a better way to this than described above ?
As far as I am aware, Asterisk has no built-in dialplan function to
allow sending of email.
>>> Nobody has any information or opinions on any of this?
Personally, I don't think MACROS are going anywhere any time soon, so I have
not bothered looking into a substitution.
As for ael; I've never used it.
Doug
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On 10/12/19 8:15 AM, Fourhundred Thecat wrote:
did you compile libmyodbc yourself ?
No,
If I recall correctly, after a lot of searching, I ran into the apt
source below and created the myodbc.list and put it into
/etc/apt/sources.list.d
cat myodbc.list
deb http://ftp.de.debian.org/debian
On 10/11/19 10:12 PM, Fourhundred Thecat wrote:
Hello,
I am trying to set up cdr logging into MariaDB through ODBC.
I have installed unixodbc unixodbc-dev and now I am struggling with
configuring /etc/odbcinst.ini
All the examples online use non-existent libraries, ie:
On my Debian Buster
Dan,
I don't run Asterisk on AWS, but I do on ESXi. Are you running a version of
Asterisk before 13? Newer versions Asterisk handle timing better that don't
require a hardware timing source.
I'm running Asterisk 13 on a small 60 phone system without issues under ESXi 6.0
Doug
--
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 would mean syncing
all sorts of data that does not need to
>>> I have updated the wiki. The script can be found within the
>>> contrib/scripts/sip_to_pjsip subdirectory of an unpacked download of
>>> Asterisk 13 and forward.
Got it!
Thanks,
Doug
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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+to+res_pjsip
It would appear that said server hosting the script is no
Maybe streaming will be helpful,
https://www.agix.com.au/streaming-internet-music-for-asterisk-10-on-hold-music/
Doug
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>>> I setup and extension to connect me with Console/Dsp. I am hearing the
>>> audio but its warbly or does not sound right. Any thoughts on what I need
>>> to do for that ?
I had that issue at a previous employer and got around it by using ALSA instead.
Doug
--
My self-compiled Asterisk also shows that speex dependencies are not installed
Speex Coder/Decoder
Depends on: speex(E), speex_preprocess(E)
Can use: speexdsp(E)
You'll need to installed the dependencies and re-compile.
Doug
--
On 7/4/19 6:40 PM, hw wrote:
This has again, and for no reason, ceased to work again after
restarting asterisk. No matter what I try, I can't create a
certificate asterisk
would verify.
Have you considered using LetsEncrypt for a valid certificate?
Doug
--
>>> We've recently replaced an old Meridian phone system (Analog) with Asterisk
>>> and signed up for Spectrum SIP trunks.
Should have included that we're running Asterisk 13, under chan_sip
Doug
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We've recently replaced an old Meridian phone system (Analog) with Asterisk and
signed up for Spectrum SIP trunks.
The service gets installed on July 8th and I was hoping somebody that may have
already gone through the process could give me some hints. I've only ever
dealt with PRIs or IAX2
core show version
Asterisk 13.26.0 built by doug @ asterisk on a x86_64 running Linux on
2019-04-05 11:41:43 UTC
Built from source,
Douh
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>>> Surely that is "call forwarding", which is quite different from either a
>>> blind or attended transfer?
That would be correct.
The forward button on the polycom phones just do a redirect to the destination
extension or external phone number.
Doug
--
We have Polycom phones (I'm using a VVX601, the destination is a VVX301).
We're also on Asterisk 13.
I forwarded my call to the VVX301 and then dialed my phones DID. The forwarded
call showed my cell phone number, so I cannot reproduce.
Doug
--
On 3/31/19 8:21 AM, Gokan Atmaca wrote:
Hello
The "res_srtp" module does not appear. How do I install it?
Are you compiling or installing from packages?
If compiling, you'll need to install the development library. Under
Debian it is libsrtp0-dev
Doug
--
>>> Does anyone have an (overhead) paging system that they like that works with
>>> SIP?
Our old phone system back ends into a Bogen AMP.
I'm in the process of replacing that system (Meridian) with Asterisk and found
that the snom PA1 works very well. If an AMP is involved, this might work.
Your IVR should only play audio prompts and only attempt to dial once a
selection has been made,
Doug
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On 3/9/19 9:56 AM, Gokan Atmaca wrote:
a) work for recording incoming / outgoing calls
b) do not work for recording internal calls
then we might be able to give you a clue what's wrong.
Hello
For example: My phone number is 1000, the other's number is 1001. These numbers
are in the same PBX
On 3/5/19 2:46 AM, Gokan Atmaca wrote:
Asterisk can send calls, but I don't get a call. What could be the problem?
[from-siptrunk]
exten => 13XXX,1,dial(${OPERATOR},20)
You are trying to match a pattern, so this needs to be
exten => _13XXX,1,dial(${OPERATOR},20)
Doug
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