Hello Federico,
Can you please review the Bug Report requirements, and submit a new bug
report for this issue?
https://docs.asterisk.org/Asterisk-Community/Asterisk-Issue-Guidelines/
Also Note:
Before filing a bug report... Your issue may not be a bug or could have
been fixed already. Run thr
Hi Dovid,
There is no default manager.conf in the 'make basic-pbx' config build.
But there is however the sample manager.conf.sample which would get
installed with 'make samples' config which has a giant security warning
at the top of the file. By default manager has enabled=no, and has a
Hi Dan,
Your best bet for looking at RTP media specifics is the standards that
define RTP.
Wikipedia has some really good resources on RTP and a list of the
various RFC standards that relate:
https://en.wikipedia.org/wiki/Real-time_Transport_Protocol
On 8/28/23 11:16, Dan Cropp wrote:
I
..snip...
etc etc
On 8/20/23 09:12, Federico wrote:
I cannot follow your instructions, because asterisk segfaults on
start. It never starts
Can you give me instruction to trap this segfault on starting asterisk?
Like gdb …..asterist –gvvc
*From:* asterisk-users *On
Behalf Of *M
Hi Federico,
Segfaults are 100% not by design. Typically if something seg faulted,
either there is a logic bug or a component mismatch. The you should
definitely be able to use more than one connection (we use multiple
connections with postgres odbc with no issue).
If Asterisk segfaults whe
On 8/18/23 12:41, Joshua C. Colp wrote:
On Fri, Aug 18, 2023 at 1:09 PM Mark Murawski
wrote:
I've seen this happen three times in the wild now. I've been
trying to
isolate the source of the issue, but so far it seems like there's not
enough debug output t
are normal for an indetermine amount of time
-
INVITE sip:+12011555432@152.X.Y.Z:5060 SIP/2.0^M
Via: SIP/2.0/UDP 192.81.237.20:5060;branch=z9hG4bK489fe.a7c59e79.0^M
From: "MARK MURAWSKI " ;tag=gK0c130ae5^M
To: ^M
Call-ID: 241982
Hi Steve,
You must be using a prebuilt system, maybe a prebuilt Asterisk-based
distribution? Asterisk does not send email by default... Almost
nothing is done by default. Things like sending email have to be
specifically configured to do so in voicemail.conf. If you don't want
to send ema
Hi Justin,
There's absolutely no detail here regarding the SIP messages going out
and back. You'll need to include the asterisk-side sip debug.
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
https://support.digium.com/s/article/How-to-collect-an-Asterisk-Debug-Capture
On 8/4/22 20:32, Jerry Geis wrote:
I am running Asterisk 13.30.0
40 core CPU (VM) VMware.
CentOS 7
32 G ram
10G vmx network
Should be plenty of room for anything...
Yes asterisk is running 270% CPU...
Is it not taking advantage of the 40 cores ?
I am bring around 300 SIP endpoints in a muted au
On 8/31/22 09:25, Antony Stone wrote:
If I simply do
Tracker="${CDR(uniqueid)}";
it works as required.
It's just not the sort of syntax I've seen in any other language, and it feels
(to me) weird.
^^^ Yup! This is what I was suggesting in my last email. Just add quotes.
Think
On 8/31/22 05:29, Antony Stone wrote:
What I am suggesting is that Tracker=${CDR(uniqueid)} should be converted
by AEL into Set(Tracker=${CDR(uniqueid)}) in order to avoid this sort of
surprise.
On the flip-side... anyone who currently relies on purely
numeric/boolean handling of the current i
On 8/30/22 17:51, Mark Murawski wrote:
On 8/30/22 12:34, Antony Stone wrote:
I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
MSet(Tracker=-1661872057.2349)
systemname is missing.
Hi Antony,
This is not a problem with MSet.
No, it is indeed the documented behaviour of
On 8/30/22 12:34, Antony Stone wrote:
I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
MSet(Tracker=-1661872057.2349)
systemname is missing.
Hi Antony,
This is not a problem with MSet.
No, it is indeed the documented behaviour of MSet "MSet behaves in a similar
On 8/30/22 11:16, Antony Stone wrote:
If I write in my AEL dialplan:
Set(Tracker=${CDR(uniqueid)});
this results in executing:
Set(Tracker=eagle.domain.com-1661872057.2349)
Just what I want.
However writing:
Tracker=${CDR(uniqueid)};
results in:
MSet(Tracke
On 8/29/22 14:00, aster...@phreaknet.org wrote:
This is a mockup of what the new-style if/else processor would output
26. NoOp(AEL IF("\${DIALSTATUS}" == "BUSY") --
extensions.ael:1405)
27. GotoIf($["${DIALSTATUS}" == "BUSY"]?28:30)
On 8/29/22 09:30, Antony Stone wrote:
It is, although there are ways I think it can be improved - I'm wondering how
best to go about proposing these.
The most obvious for now are:
- please can "a=1;" be converted to use Set() instead of MSet() (especially
since MSet is officially deprecated)
On 8/29/22 10:15, Antony Stone wrote:
But! What specific reason do you have for wanting Set() instead of
MSet() for all assignments that can't be otherwise just written as an
in-line Set() instead?
I *am* currently writing inline Set() everywhere, but surely the syntax "a=1;"
instead of "Set(a=
On 8/29/22 09:53, Antony Stone wrote:
On Monday 29 August 2022 at 15:35:09, Joshua C. Colp wrote:
MSet is not deprecated.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MSet
includes the sentence "MSet behaves in a similar fashion to the way Set worked
in 1.2/1.4 and is thu
On 8/29/22 08:48, Mark Murawski wrote:
On 8/29/22 08:31, Antony Stone wrote:
Hi.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Originate
I need to use Originate() in a dialplan, pointing to another location
in the
same extension of the same context, so for example
On 8/29/22 08:31, Antony Stone wrote:
Hi.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Application_Originate
I need to use Originate() in a dialplan, pointing to another location in the
same extension of the same context, so for example:
Originate(Local/${Dest}@Dialout,exten,${CONTEX
On 3/1/22 05:59, Karsten Wemheuer wrote:
Am Dienstag, dem 01.03.2022 um 06:37 -0400 schrieb Joshua C. Colp:
On Tue, Mar 1, 2022 at 6:14 AM Karsten Wemheuer wrote:
Hi *,
i am currently trying to migrate from chan_sip to pjsip. I am using
Asterisk version 18.10.
In chan_sip information about t
Hi Antony,
NOW is not a variable...
In the majority of cases (the exceptions are things like CUT)...
variables are utilized by ${}
If NOW was a variable you would see it written as ${NOW}
The word NOW is actually not special. Deep in the Asterisk source (if
you are curious), the flow is th
If you're executing /usr/bin/rm directly, shell aliases will have no effect.
On 1/11/22 11:29, Antony Stone wrote:
On Tuesday 11 January 2022 at 17:20:44, Michael Englehorn wrote:
If you're on RHEL or CentOS or one of its descendants,
Oh, now that reminds me that those systems also tend to a
Hi Daniel,
This is a production server which is running well over years (asterisk
11-13-16) and this happend with the latest version. Only valid option
you gave is the core show locks. I ask the list before opening a bug
report, as usually.
Please don't let the fact that the system has bee
Hi,
1) You should change your name on your email client so it doesn't say
"Administrator"
2) Please follow the instructions at
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source
3) Compile with DEBUG_THREADS and DONT_OPTIMIZE, but note this will
incur a performance
On 10/17/13 23:06, John T. Bittner wrote:
Today I was hacked but caught it very quickly. This is the weird part,
they hacked an IP Auth based account by simply knowing the account name.
How is this possible? I am running Asterisk 11.5.0. Now it’s my fault I
used a dictionary based account name
On 12/27/2012 07:36 PM, Ron Wheeler wrote:
On 27/12/2012 3:14 PM, Carlos Alvarez wrote:
On Thu, Dec 27, 2012 at 12:46 PM, Ernie Dunbar mailto:maill...@lightspeed.ca>> wrote:
This past holiday weekend has resulted in some real groaners when
it comes to bugs in our dialplan, making obviou
Asterisk 1.8.5
Polycom Bootrom 4.4.0
Polycom spip 4.0.1
They are all sip devices talking to each other.
Polycom phone A puts polycom phone B on hold, Phone A tries to unhold
the caller but the line button is still flashing on hold like nothing
happened.
All sip peers are directmedia/canreinvi
nfig.c:1154 process_text_line: No
'=' (equal sign) in line 136 of /etc/asterisk/extension-operator.conf
[Apr 3 20:12:17] WARNING[28078]: pbx_config.c:1499 pbx_load_config:
==!!== Unknown directive: s at line 135 -- IGNORING!!!
thanks for your help
olivier
2011/4/3 Mark Murawski:
In that si
In that situation, I've had to do a pickup macro that kind of "primes"
the audio.
Dial(SIP/MyOperator/${NUMAPPEL},180,rtU(start-audio))
context start-audio {
s => {
Playback(silence/1);
}
}
The above might help... What it does is plays an audio track on the
callee's channel (SIP/MyOpe
/global/documents/support/setup_maintenance/products/voice/spip_ssip_vvx_Admin_Guide_SIP_3_2_2_eng.pdf
On Fri, Mar 25, 2011 at 6:33 PM, Mark Murawski
wrote:
Sorry for the crosspost. This was supposed to be on -users
I know some of you are polycom gurus...
Anyone know how to remove transfer from a poly
Sorry for the crosspost. This was supposed to be on -users
I know some of you are polycom gurus...
Anyone know how to remove transfer from a polycom 33x phone? We've set
allowtransfer=no, but we would like to remove a polycom soft key as well.
--
___
See pickup macros... the U option to Dial.
On Sun, 23 Jan 2011, Michelle Dupuis wrote:
I need to do some stuff in the dialplan BEFORE either leg of the call is
started...
--
_
-- Bandwidth and Colocation Provided by http:/
On 01/17/2011 08:26 PM, Matt Riddell wrote:
On 17/01/11 4:29 PM, jon pounder wrote:
Surely there is some mail client smart enough to be able to flip around
the levels of indenting so most recent is top or bottom.
If not quit bitching and make one - I will continue top posting since I
don't seem
On 01/16/2011 10:28 PM, Mark Murawski wrote:
We obviously have all our own opinions about being on top or bottom. And
it boils down to personal preference obviously.
And it looks like I top posted, heh. I just usually hit reply and start
typing, the default is top.
I guess I go both ways
We obviously have all our own opinions about being on top or bottom.
And it boils down to personal preference obviously.
I think in all cases, top posting is by far superior. But I think the
battle will continue ad infinitum.
One, because of speedups in finding the most recent content which
Seconded. Although I've succumbed to bottom posting on occasion
when following the convention of the ongoing thread.
On 01/14/2011 07:42 PM, Don Kelly wrote:
Bruce et al…
I’m posting a
new thread with
rride some of the sip.cfg settings in the polycom dir with:
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
*** Need help? Contact supp...@amplisys.ca <mailto:supp...@amplisys.ca> ***
On 2011-01
---
Sebastien Thomas
Amplisys Inc. - Digital Telephony Integration Specialists
T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS
On 2011-01-13, at 1:32 AM, Mark Murawski wrote:
Would anyone happen to have some examples of polycom configs, specifically the
650 with sidecar for blf.
I
Would anyone happen to have some examples of polycom configs,
specifically the 650 with sidecar for blf.
I have the asterisk side all configured since I've set up blf with other
types of phones, but I'm missing the polycom side.
I've put together a -directory.xml, and the sidecar now lists
n
On 01/05/2011 01:51 PM, Tom Rymes wrote:
On 01/05/2011 7:50 AM, Andy Graybeal wrote:
We've got two noisy kitchens that need to talk back and forth.
Andy,
Why, exactly, are you trying to combine an inter-kitchen intercom and
your phone system? Might it make more sense to have a non-phone-base
Looks like your telco is sending you polarity reversal on sending you a
call. Which is one of the types of setups for analog lines.l
From your console output it looks like the call was handled just fine
other than the 'weird event' notification, which I'm not familiar with.
On 01/05/2011 1
o treat this issue?
>
> Until now, we use asterisk for routes call in our office. Now we are
> thinking in integrate asterisk with our app.
>
> Thanks once again!
>
> Regards
>
> *Sidarta Oliveira*
>
>
> --
Local channels behave like an endpoint. So instead of a sip phone
picking up the call, asterisk is picking up the call.
Instead of someone speaking into a sip phone, asterisk can play tracks,
or record digits, etc.
You need to make sure that the call does not end before you're done with
your
Are you using originate? Check your originate timeout.
Are you limiting your call length on Dial()... check your L options.
Asterisk will send a BYE if it hits an internal timer that's set to
destroy the call at a specific time.
For instance... this is almost guaranteed to cause problems
Actio
From: "Paul Belanger"
Sent: Thursday, October 14, 2010 6:43 PM
To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
Subject: Re: [asterisk-users] Audiocodes firmware
On Thu, Oct 14, 2010 at
Belanger wrote:
> On Thu, Oct 14, 2010 at 3:25 PM, Mark Murawski
> wrote:
>> Does anyone have links to the most recent audiocodes firmware?
>>
> Why not contact Audiocodes?
>
--
_
-- Bandwidth and Col
Oh right...
MP-118
Thanks.
On 10/14/2010 03:38 PM, Bryant Zimmerman wrote:
For which device models?
From: "Mark Murawski"
Sent: Thursday, October 14, 2010 3:26 PM
Does anyone have links to the most recent audiocodes
firmware?
--
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