On Wed, 16 Aug 2023, Federico wrote:
But now I upgraded to Asterisk18 and there is no longer a local channels
Are app_originate.so and res_clioriginate.so loaded?
--
Thanks in advance,
-
Steve Edwards sedwa
:
On Sat, Jun 17, 2023 at 7:48 PM Steve Matzura wrote:
OK, this is how I thought it's supposed to work. It just
confounded me why the book would say the Playback() and
Background() syntax were the same, then in the very next paragraph
give an example that belied that claim.
The syntax
OK, this is how I thought it's supposed to work. It just confounded me
why the book would say the Playback() and Background() syntax were the
same, then in the very next paragraph give an example that belied that
claim.
On 6/17/2023 1:46 PM, Doug Lytle wrote:
On 6/17/23 08:47, Steve Matzura
Doug,
This is where the weeds start growing.
On 6/17/2023 4:55 AM, Doug Lytle wrote:
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.
Understood. From the book:
You all know the story--give the customer/client what they ask for, and
if they like it, they'll be back for more. Such is just so with my
one-trick-pony answering-machine project. Now the other two musicians in
my virtual band want the following capabilities:
1. The ability to dial the main
I'm setting up voicemail on my answering-machine project.
Since the directory for voicemail messages for an extension doesn't
exist until there's a message to be saved therein, how can I create a
custom greeting since it goes in that directory? That's what it sounds
like the book is telling
On 5/28/2023 2:27 PM, Naveen Albert wrote:
However, you can also pass audio without supervising (early media).
You typically need to Progress() first to allow this, e.g. for SIP, or
audio won't pass at all.
...
If you want it to ring once and do something else, you could simply do:
exten
On Sun, 28 May 2023, Steve Matzura wrote:
It's probably eight or nine years old now, an ASRock motherboard with I
don't even know what on it in the way of processor speed or power. I
should probably pick up another machine but I can't justify the expense
because it's only for play, FTP
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--the provider or Asterisk? If it's the DID provider, I'll
work on that with them; if it's Asterisk, I didn't find anything
anywhere that looks like
On 5/28/2023 6:19 AM, aster...@phreaknet.org wrote:
A great reason to avoid Asterisk packages and compile from source
instead. You'll save yourself a lot of headaches.
That's how I started, by trying to build version 18 from source. It
failed. Colossally. The compile of sources would run
On 5/27/2023 11:40 AM, aster...@phreaknet.org wrote:
Relative paths are relative to your language-specific directory.
Ya know, that's the one thing I didn't do was test Playback before
copying the sound files out of /usr/share/asterisk/sounds/en_us into
/var/lib/asterisk/sounds--I don't
Acording to the book, I'm supposed to put things into what Asterisk
thinks is its default audio file location, /var/lib/asterisk/sounds, and
I'm supposed to be able to create a custom directory off of that path
and use it in a relative-syntax way in the Playback directive, like so:
...
SIP.
On 5/27/2023 10:23 AM, Steve Matzura wrote:
Sean,
I'll take that under advisement, but Doug swears by IAX, I tried it,
it worked, so until things break and break bad, I'll stick with that
and try the recommended remedy, now recommended by two people.
On 5/26/2023 8:08 PM, Sean Bright
., Steve Matzura wrote:
And I think they're both small.
[May 23 18:34:12] NOTICE[46582]: res_pjsip_session.c:3968 new_invite:
voipms: Call (UDP:208.100.60.12:5060) to extension 's' rejected
because extension not found in context 'voipms-inbound'.
Steve,
In your voip.ms console, go to Account
Sean,
I'll take that under advisement, but Doug swears by IAX, I tried it, it
worked, so until things break and break bad, I'll stick with that and
try the recommended remedy, now recommended by two people.
On 5/26/2023 8:08 PM, Sean Bright wrote:
On 5/26/2023 5:41 PM, Steve Matzura wrote
Doug from this list got me to change my connectivity to my DID provider
from SIP to IAX, and bingo, it all just worked instantly.
For my next trick: setting up voicemail. The book does it all with smoke
and mirrors (SQL), but I'm fresh outa those, so I'll be doing it the
old-fashioned way,
This was supposed to go to the list.
I am now thoroughly confused.
In the [voipms] stanza where endpoint is defined (type=endpoint),
everything points to voipms. But in the [yealink] stanzas, I tried
pointing everything
to Steve, one item at a time, then both of them, and nothing changed
And I think they're both small.
Solved: tcpdump showed no packets coming in, so I went to my DID
provider's Website to discover to my intense embarrassment that the DID
number had been set up forwarded to their voicemail. I got egg on my
face for this one. I changed that setting to SIP/IAX
On Tue, 23 May 2023, Steve Matzura wrote:
The "Definitive Guide" shows everything about adding phones as SQL
statements...
I'd look for another guide.
--
Thanks in advance,
-
Steve Edwards sedwa...@se
On Tue, 23 May 2023, Steve Matzura wrote:
...when I dial my number from a phone on the Internet or any phone
outside my LAN, Asterisk does not respond in any way, which means
somehow my system is not picking up the fact that there's an incoming
call to it.
Or that you are not receiving any
(see below).
Here's how I set it up in pjsip.
[yealink]
transport=udp
type=auth
auth_type=userpass
username=Steve
password=Steve
[yealink]
type = endpoint
transport = transport-udp
context = phones
disallow = all
allow = ulaw
; allow=g729 ; uncomment if you support g729
auth = ye
the discussion too)
-Original Message-
From: Steve Matzura [mailto:s...@noisynotes.com]
Sent: Monday, May 22, 2023 12:15 PM
To: TTT
Subject: Re: [asterisk-users] Ready to throw up my hands in defeat
Thanks. Further reading and digging did in fact prove out that the RTP
is a lot of what's
I am not comfortable with admitting this on a public userlist [;-)] but
after over forty years in software development and manual-reading and
-interpretation, I've finally hit one that I can't get past.
I've mention previously that I worked with Asterisk in older days--like
in around
if this is
desired behavior or not, but it's how pjsip seems to work.
On 4/6/23 2:54 PM, Steve Sether wrote:
We've been using Asterisk 16 for a while now, and tried turning on
send_rpid = yes in my pjsip config for end points. This solves a
problem we're having where attended transfers aren't
are the
'method,' 'tos' and 'cos' keywords, which are commented out in your
instructions?
Otherwise, the rest is quite clear.
On 4/8/2023 12:35 PM, Michael Maier wrote:
Hello Steve,
use the following configuration for the transport and bind this
transport to the trunk:
[transport_name]
type=transport
I want to configure communication with my phone provider using TLS for
all the obvious reasons. Since I'm behind a firewall, I'll be needing to
do it with NAT. There are examples of UDP plus NAT in pjsip.conf, but
none for TLS plus NAT. Would it be correct to set up the TLS transport
stanza to
Sorry, meant version 16, like the book. Sure would prefer 20.
On 4/6/2023 3:30 PM, Steve Matzura wrote:
It appears I have bigger problems heretofore unknown. I've gone
through this several times today since I last wrote, and the
phreaknet-run build failed every time, but each time
We've been using Asterisk 16 for a while now, and tried turning on
send_rpid = yes in my pjsip config for end points. This solves a
problem we're having where attended transfers aren't updating the
CallerID when the transfer is complete (it would show the callerID of
the party attempting the
want something easy to use out of the box, install the
FreePBX distro.
Given that Steve originally said "I've been using Asterisk, including
administering and maintaining it, in some aspect since 2003, but this is the
first time I have attempted a from-scratch installation and setup on my ow
system, I'm just going to leave it that way and work with it as it is.
On 4/6/2023 10:35 AM, Antony Stone wrote:
On Thursday 06 April 2023 at 15:48:24, Steve Matzura wrote:
this is the first time I have attempted a
from-scratch installation and setup on my own.
..
Then the weeds
I've been using Asterisk, including administering and maintaining it, in
some aspect since 2003, but this is the first time I have attempted a
from-scratch installation and setup on my own. I'm following the
instructions in the ePub edition of the book "Asterisk, the Definitive
Guide, Fifth
user authentication. This response
is issued by UASs and registrars.[1]: §21.4.2"
My guess would be a user or password mismatch.
Are you using SIP or PJSIP?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
dvance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided b
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281--
_
--
?
AFAIK, # is it.
I use 'wait for digit' in a loop to accumulate digits so I can terminate
entry based on the number of digits or a specific key.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1
We're having a problem where Asterisk 16 refuses to play voicemail
recordings and greetings stored in wav49 format. It throws an error
similar to the following:
2022-01-27 11:31:37 format_wav.c: Not a supported wav file format
(49). Only PCM encoded, 16 bit, mono, 8kHz/16kHz files are
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth
, hangup()
Hopefully somebody else has a more elegant solution.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
On Wed, 5 Jan 2022, Steve Edwards wrote:
same = n, set(LAST-CONTEXT=${context}
Double damn. I munged the case on ${CONTEXT}. I give up for today :)
--
Thanks in advance,
-
Steve Edwards
On Wed, 5 Jan 2022, Steve Edwards wrote:
same = n, set(LAST-CONTEXT=${context}
Damn. forgot the closing parentheses :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281--
_
-- Bandwidth and Colocation Provided by http://www.api
would ideally like to do it in one line.
1) gotoif()
2) gosub()
3) AEL
gosub() is probably 'cleaner' and more maintainable than gotoif(). AEL is
good but sometimes fragile.
--
Thanks in advance,
-----
Steve Edwards sedwa...
On Wed, 22 Dec 2021, Steve Edwards wrote:
same = n, set(ARRAY(foo1,foo2,foo3,foo4)=1,2,3,4)
Just to be clear...
The use of sequential ascending numbers in all of the examples should not
be construed as having any meaning. You could just as easily have:
same = n, set(ARRAY
'ARRAY()' :)
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
languages exhibit similar behavior?
I have no specific knowledge of Python/AGI. Sorry.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com
On Fri, 12 Nov 2021, Steve Edwards wrote:
I prefer to do database work in an AGI. I find quoting within the database to
be obtuse and fragile.
s/database/dialplan/g
--
Thanks in advance,
-
Steve Edwards sedwa
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided
and then hangs up without even bothering to
answer).
Any suggestions welcome :)
How about creating a call file in the h extension?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth
.
If you enable SIP debugging (and bump up debug and verbose), is the delay
between when you dial and the INVITE is displayed or is the delay between
the INVITE and subsequent steps in your dialplan.
--
Thanks in advance,
-
Steve
(the
BIN) and the last 4 digits and replace the rest with x. We used to call
the result a 'span.' I have no idea if this is current practice.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice:
er you 'set autohangup x' just set 'TIMEOUT(absolute)=${EPOCH}+x.'
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/
> agi_version: 13.14.1~dfsg-2+deb9u4
AGI Tx >> agi_callerid: 55
AGI Tx >> agi_calleridname: Steve Edwards
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: *
AGI T
789'
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and C
On Thu, 6 May 2021, Jonathan H wrote:
"bumps up the outgoing volume to +7"
I use 'normalize --amplitude=-22dB" to adjust volume levels to consistent
levels.
--
Thanks in advance,
-----
Steve Edwa
On Fri, 26 Feb 2021, Dovid Bender wrote:
Steve,
What language are your AGI's written in? I have been using PHP for a long time
and every time it's launched there seems to be a run on the CPU. I wonder if I
would be
better off using Python or something other than PHP.
C.
--
Thanks
.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
We have an auto-pause feature where agents are paused after a call, and
manually un-pause when they're finished with wrap-up. This worked
perfectly in Asterisk 11.
We've recently switched to Asterisk 16, and we now occasionally hear
reports of users saying a call rang-through after the
().
Almost always exit().
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
out of service copper number,
555-555-.
I'm all for the discussion, but can you start a new thread so we don't
keep associating the innocent party (the OP) with this spammer.
--
Thanks in advance,
-
Steve Edwards
r any version of Asterisk, if
interested contact me at venefax at the Google mail service."
Fixed. If you're going to post a commercial solution on a non-commercial
forum, at least be up front about it.
--
Thanks in advance,
-
xes using cards so I can't test.)
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244
est
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
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provid
/res_pjsip/pjsip_configuration.c
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
only allowed up to 31 character passwords.
You may find it useful to use tcpdump with '-w' to write the packets to a
file and then analyze with sngrep.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
We get some noise in our Asterisk error file generated by scanners
sending invalid invites. Example below (details removed)
[2020-12-0702:53:30]ERROR[23370]pjproject:sip_transport.c Error
processing 559 bytes packet from UDP ***
:PJSIPsyntaxerrorexceptionwhenparsing'Request
Joshua is lieing ASSHOLE
Sent from my iPhone
> On Sep 30, 2020, at 7:08 AM, Joshua C. Colp wrote:
>
>
>> On Wed, Sep 30, 2020 at 9:06 AM sergio wrote:
>
>> On 30/09/2020 14:59, Joshua C. Colp wrote:
>>
>> > latest version of 16 on Ubuntu
>>
>> 16.12.0~dfsg-1 ?
>
> I don't use packages.
On Fri, 11 Sep 2020, sean darcy wrote:
I'd like to get an alert if a call fails to authenticate:
if "Failed to authenticate" then
mail someone the source ip
endif
How about fail2ban?
--
Thanks in advance,
-
On Sun, 12 Jul 2020, Steve Edwards wrote:
So this is a provider issue, not an end user issue and 'June 30, 2021'
doesn't sound like 'soon.' If this is legit, why haven't my providers
said squat?
Seems one of my providers, Vitelity (iax.cc to us old timers), when asked,
is not panicking
is
'token' add any value?
What am I missing?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwar
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided
On Fri, 12 Jun 2020, Jerry Geis wrote:
Any chance you can configure the speaker to syslog to your host so you may
get a clue why the speaker is rejecting?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and
as the caller hangs up), and then
rewrite and reload again when there's a new caller.
How about ARA to configure MOH and then just update the database.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice
ly from the
Ethernet on the speaker to a NIC on the computer? It doesn't matter, just
curious :)
The only thing that will tell you what is going on is the packets. Crank
up 'sip set debug on' and see if that yields a clue.
--
Thanks in advance,
---
On Wed, 3 Jun 2020, Fourhundred Thecat wrote:
On 2020-06-03 17:21, Steve Edwards wrote:
How about:
syslog.local0 = error,verbose,warning
no debugging detail.
syslog.local0 = debug,error,verbose,warning
include debugging detail
about:
syslog.local0 = error,verbose,warning
no debugging detail.
syslog.local0 = debug,error,verbose,warning
include debugging detail.
--
Thanks in advance,
-
Steve E
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided by http://www.api
On Wed, 27 May 2020, Saint Michael wrote:
We are in the business of...
Then this probably should have been posted on -biz.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
ne") == 0) {
rotatestrategy = NONE;
} else {
fprintf(stderr, "Unknown rotatestrategy: %s\n", s);
}
So, backport or upgrade?
Also, inquiring minds want to know why the enum is in powers of 2? It's
not like we can set
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
--
[general]
rotatestrategy = none
[logfiles]
/tmp/ast-log-test =
debug,dtmf,error,event,notice,verbose,warning
; (end of /etc/asterisk/obl/logger.conf)
--
Thanks in advance,
-
On Fri, 24 Jan 2020, Steve Edwards wrote:
2) How about doing 'GET FULL VARIABLE' in your Perl script?
Sorry. After a couple more cups of tea I think this was a bit vague.
Try whatever call/method in your library that does 'GET FULL VARIABLE' on
'${PJSIP_HEADER(read,P-Asserted-Identity
ll header? Try 'verbose(PAI
= ${PAI})' or something similar.
2) How about doing 'GET FULL VARIABLE' in your Perl script? You can set
the channel variable PAI in the AGI if needed back in the dialplan.
--
Thanks in advance,
---
Sipura 3000?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
, increase
consistency, and reduce maintenance.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwar
because I'm just that kind of guy :)
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve
; 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/
On Fri, 7 Jun 2019, David Cunningham wrote:
We're using Perl and so far I haven't found an equivalent there.
On Thu, 6 Jun 2019, Steve Edwards wrote:
I'm not much of a Perl programmer...
But you should never turn down an opportunity to develop your skills :)
Try something like
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4
On Fri, 7 Jun 2019, David Cunningham wrote:
We're using Perl and so far I haven't found an equivalent there.
On Thu, 6 Jun 2019, Steve Edwards wrote:
I'm not much of a Perl programmer...
But you should never turn down an opportunity to develop your skills :)
Try something like
}
Looks like agi_environment.result is your Huckleberry.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244
ful (long) options and are not dependent upon passing arguments in
a particular order.
--
Thanks in advance,
-----
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/stev
in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
.
The only caveat is to not interact (stdin/stdout) with Asterisk until
'stream file' in the thread completed.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
is not a POTS feature.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
1 - 100 of 8788 matches
Mail list logo
