Thank you Thank you Thank you! I changed it to: exten =>
s/555333,1,Gosub(subBusy,s,1()) and it now works like a charm. Really
appreciate the help!
El sáb, 20 nov 2021 a las 10:55, escribió:
> On 11/20/2021 11:51 AM, Richard Reina wrote:
> > Since Macro is deprecated
Since Macro is deprecated I am trying to eliminate it from my diaplan. I
believe I have successfully done so in the example below.
; dial an internal extension
exten => 101,1 Macro(ext,100,Dahdi/15)
TO:
exten => 101,1,Dial(Dahdi/15,30)
So far it seems to work. However I also in my dialplan
There are semicolons in the useragent string you are trying to set. If
that is the exact dialplan line then
those semicolons are being seen as a start of a comment.
Richard
On Mon, Dec 14, 2020 at 12:25 PM Jonathan H wrote:
> All my other CURLOPT settings like timeout work f
I am getting zero interrupts for a new Digium TE134 Card on a new brand new
Dell T40 server with the latest BIOS. Is there something that I am missing
or is the card not compatible with Dell servers?
(cat /proc/interrupts ; sleep 1 ; cat /proc/interrupts) | grep -i wcte13xp0
16: 0
use it in the dialplan?
>
> Thank you in advance
>
The function CHANNEL(callid) returns exactly what you want.
See https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+Function_CHANNEL
Richard
--
_
-- Bandwidth and Colocation Pro
Argh. That was for chan_pjsip and you are using chan_sip. Be aware that
chan_sip is effectively dead.
Richard
On Thu, May 14, 2020 at 9:50 AM Richard Mudgett wrote:
> The other end is sending g729 even though it was not negotiated. The
> other end should not do this and it usually
The other end is sending g729 even though it was not negotiated. The other
end should not do this and it usually seems that the other ends that do
send g729.
This was recently fixed. See
https://issues.asterisk.org/jira/browse/ASTERISK-28139
Richard
On Thu, May 14, 2020 at 1:11 AM John Hughes
I am developing apps using ARI which need suppression of DTMF tones in the
audio, and I have been told (back in December) that asterisk depends on SIP
providers to suppress DTMF tones in the audio stream.
Having sorted out my ARI code to suppress DTMF as I wanted, it turns out that
SIP
l the test
extensions to see what is on the channel's hangup handler stack while the
channel is in the Echo application by using the command line commands
mentioned on the wiki page.
Richard
On Mon, Feb 3, 2020 at 7:26 PM David P wrote:
> Please point me to samples of popping and wiping hangup
1 root root 887 nov. 18 20:46 asterisk.key
> -rw--- 1 root root 2111 nov. 18 20:47 asterisk.pem
>
I'd say that asterisk running as the asterisk user has no permission to see
the .pem file as only root can see it.
Richard
> -rw--- 1 root root 161 nov. 18 20:46 ca.cfg
> -rw--
P MESSAGE requests
in the dialplan. It cannot be hung up.
Richard
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New
ions?
>
You must have multiple patterns to match the various starting sequences you
receive.
One that begins with +
One that begins with 1
One that is for a 10 digit number
Richard
--
_
-- Bandwidth and Coloca
uffers => 12,half
> channel => 49-53
>
>
It will not work. This is a limitation of analog lines. In this case your
Asterisk box is
pretending to be a POTS phone. Phones do not tell the PSTN who they are as
the PSTN
already knows who they are.
Richard
--
__
message is not an error. It is a verbose log stating what it did. It
is a result
of you telling the Dial application to block the initial connected line
update because
you set Dial's 'I' option flag.
Richard
--
_
-- Bandwidth a
;
>
> do you think if this can be bug?
>
It is not a bug. The contact has been "created". It will stay in that
state unless
you are also going to qualify the endpoint. Asterisk 16 simply renames the
state to
"NonQualified" to be more explicit.
Richard
On Tue, Oct 23, 2018 at 5:07 PM Jonathan H wrote:
> Thanks Richard - any idea if these matter? And how to stop the errors:
>
> cdr_sqlite3_custom declined to load.
> cel_sqlite3_custom declined to load
> pbx_ael declined to load
>
> Standard 16.0 build, just updated a
; is needed when using chan_sip and
res_pjsip_transport_websockets on
; the same system.
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Astricon i
ef="
> http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-15-current.tar.gz
> ">download now
> But when the page is loaded it downloads the new version.
>
Should be fixed.
Richard
--
_
-- Bandwi
es instructions on how to build them.
https://wiki.asterisk.org/wiki/display/AST/Installing+Asterisk+From+Source
>
> 2. On a general point of view, is collectd daemon extended with a statsd
> plugin able to collect Asterisk Statsd statistics ?
>
I don't know.
Richard
>
> [
xist
> ?
>
The reload message is incorrect when the statsd.conf file has not changed.
I have just
put up a patch on gerrit [1] to fix it. There was another fix [2] made
about a week ago that fixed
a more general problem that affected the reported reload status of any
module if it failed.
Richard
[1
for chan_sip's behavior and chan_pjsip's
asymmetric_rtp_codec=no option is because phone DSP's can only handle
a single codec at a time. Technically if the peer's INVITE offered five
and we responded with three the peer should immediately renegotiate to
narrow the
codec choice to one.
Richard
the
uniqueid of the oldest associated channel
within the Asterisk box. Uniqueid's are unique within an Asterisk box and
can be made unique across Asterisk boxes
by optionally adding the host name.
Richard
--
_
-- Bandwidth and Colo
pairs. When you dial a local channel it is
the ;1 channel that acts as an
outgoing channel and the ;2 channel executes dialplan and acts as an
incoming channel. This is the
initial role between the two channels in a local channel pair. In the
transfer scenario I describe above
(the an
ll shows that DYNAMIC_FEATURES is set. It's just not accessible.
>
> Any thoughts?
>
It likely depens on how you are doing the attended transfer. Via DTMF?
Via SIP or channel technology protocol?
Does the Agent B channel have the DYNAMIC_FEATURES channel variable set on
it?
Rich
core set debug atleast X
That is why the debug level does not go down. Another thing is that the
debug level
is global to the system. Thus if you set the level in one connection it
affects all
connections including future ones. The verbose level is per connection.
Richard
--
___
rmation goes away when
the local
channels optimize out.
The Dial 'L' option currently puts state on the caller and called channels
depending on which
features are configured (who hears things). If you set the verbose level
to 4 you get information
in the log about that.
Richard
[1] https:
se loggers, they are now only logged as debug messages.
Richard
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New to Asterisk
ely different "asterisk" ELF binary each time I
> recompile asterisk, according to checksum?
>
> Can someone shed light...
>
A timestamp is added to the version string when you build Asterisk. Thus
every
time you recompile Asterisk you get
d from stdin.
( mailcmd < temp-email-body-file ; rm -f temp-email-body-file ) &
Richard
--
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owChannels
Or you can go to the wiki
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+AMI+Actions
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+ManagerAction_CoreShowChannels
Richard
--
_
-- Bandwidth and Colocation Pr
nywhere. I
> am using Asterisk 15.4.1.
>
You have to start asterisk with the -g option to make asterisk create core
files.
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace
Richard
--
_
-- Bandwidth and Colocation
The most-up-to-date and accurate option documentation for your Asterisk
version will be what is installed
online with your Asterisk installation. In this case CLI "config show help
res_pjsip global endpoint_identifier_order",
and "core show help pjsip show identifiers".
is to define a tone event
called alias with a tone cadence of "gb".
Richard
On Mon, Apr 23, 2018 at 1:08 AM, Patrick Wakano <pwak...@gmail.com> wrote:
> Hello list,
> Hope you all doing fine!
> I've tried to use the 'alias' directive in the indications.conf file but
>
are used to create the specified channels on
the next line.
channel=1-15,17-31
; Any options set AFTER the channel line above DO NOT affect those channels.
context=other
Richard
--
_
-- Bandwidth and Colocation Provided by http://ww
On Tue, Apr 3, 2018 at 4:57 PM, Matt Fredrickson wrote:
> On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield
> wrote:
> > In article
n cdr-scv/Master.csv
> file ?
>
Before CDRs get written to the back ends (i.e., permanent storage) they are
in memory data structures. Where else could they be before getting written
to the back ends?
Richard
--
_
--
r thread to write to
the back ends.
You need to be using at least v13.19.1 or v15.2.1 to also have some CDR
performance enhancements to
help CDR processing of call events.
For information about ODBC connection pooling performance problems see [2].
Richard
[2] http://blogs.asterisk.org/2016/
aor-single-reg](!)
> type=aor
> max_contacts=20
>
> [1001](endpoint-basic)
> auth=auth1001
> aors=1001
>
> [auth1001](auth-userpass)
> password=password123
> username=1001
>
> [1001](aor-single-reg)
>
>
> Extensions.conf
>
> [from-twilio]
&g
mixing. ConfBridge does not
need DAHDI since
it does its own mixing in Asterisk.
As for timing sources you have several to choose from of which DAHDI is one
of them. See
menuselect res_timing_xxx modules. Timing is really only needed when
playing back sound
prompts when nothing is
I have an old setup based in Asterisk 1.8. The carrier is accepting Ulaw in
the initial invite, but immediately the call is established they send a
re-invite to change to Alaw. This doesn't get transcoded and the user gets
no audio from after the re-invite
Is/was this a problem for asterisk
> >> If you can provide details, even vague ones, about how you did it, I
> >> can update the WMM package.
> >
> > See http://asterisk.gnat.com/meetme.tgz
> >
> > That's a gzipped tar of our working directory plus the relevant parts of
> > extensions.conf. I xxx'ed out phone numbers and Google
ed to the queue which should
> distribute to connected agents. is this possible on teh actual
> app_queue or we would need to implement it using ARI.
>
> Thanks in advance.
>
You need to use app_queue[1] with app_agent_pool[2][3][4] for your agents.
Richard
[1] https://wiki.aster
iler
can optimize out variables that could make understanding what is going on
harder.
So it depends upon what happened if an optimized backtrace can help find
the root cause
or not. It is up to you whether you want to run in production with an
optimized build or not.
I also recommend
nager API
> Can anybody explain how the native format is chosen in these cases?
>
Version 13.1 is a very old version of Asterisk 13. The current version of
Asterisk 13 is 13.18.2.
I also recall an issue where local channels tended to use slin192 when
there was no need. Howev
On Tue, Nov 21, 2017 at 5:04 AM, Benoit Panizzon <benoit.paniz...@imp.ch>
wrote:
> Hi Richard
>
> Thank you
>
> > You need to set more redirecting information [1].
> >
> > In sip.conf send_diversion=yes needs to be in effect. You also need
> > to setup
&
id information (at least the from number) to indicate where
you
are redirecting from. You should also increment the redirecting count.
chan_pjsip has the same requirements. pjsip.conf send_diversion=yes needs
to
be in effect and you also need to setup the from party id information.
Richa
k that case was
because the device was converting ISDN to SIP. I do think that the devices
that don't stop local ringback in favor of the incoming RTP stream following
the 183 are broken. Unfortunately it is something that is out of your
control.
Richard
--
_
l suppress them on the console.
Richard
--
_
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Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
he dial
application. This dialplan
needs to be able to distinguish between the two channels and act
accordingly. Using
the F() option with a dialplan location is the simplest way to distinguish
between the
two channels.
Richard
--
_
> If you can provide details, even vague ones, about how you did it, I
> can update the WMM package.
See http://asterisk.gnat.com/meetme.tgz
That's a gzipped tar of our working directory plus the relevant parts of
extensions.conf. I xxx'ed out phone numbers and Google interface data.
This
> I have a very old server that is used only for conferences on
> Meetme. To manage the conference rooms we use Web Meetme. Now it is
> time to upgrade everything but since Meetme is no longer available I
> need to find a replacement GUI to manage the conference rooms. Anyone
> know a
quot;
based off of the currently checked out "master" branch. This "13" branch
is just
another branch of master and not a real 13 branch. When you tried to put
it up for review
to the real 13 branch using "git review 13", git tried to merge your master
"13" br
tell pjsip how to send
and receive SIP messages. The provided samples may or may not apply to
your particular network.
* The name of a transport section is completely arbitrary. What makes it a
transport section is the "type=transport" line.
[1] https://wiki.asterisk.org/wiki/di
e more than 4k simultaneous calls.
>
* There is no user configurable option to change the excessive ref count
trigger value. However, you could change the EXCESSIVE_REF_COUNT define
value in the main/astobj2.c file and recompile.
Richard
--
We're experimenting with using Asterisk (14.6.0) for video conferences.
This test has three endpoints, a Polycom Trio with its video accessory,
and two desktops running Linphone. The video is all H.264. We're using
Opus for audio on the Linphone Windows desktops and have tried both
G.722 and
On Mon, Aug 28, 2017 at 6:35 PM, Richard Kenner <ken...@gnat.com> wrote:
> I've had two Asterisk crashes today that seem to be caused by errors
> where chan->tech_pvt is pointing to something that can't be deallocated
> and I think I see a reference count bug in
I've had two Asterisk crashes today that seem to be caused by errors
where chan->tech_pvt is pointing to something that can't be deallocated
and I think I see a reference count bug in the above function.
It contains:
if (data->chan_old_vsrc) {
lls/sec and the calls lasting 8 seconds that comes to 4000
active channels. Hitting the FRACK would result in an average of 25
references to the format per channel. This is a simplistic calculation as
there are going to be some references that have nothing to do with a call.
The number o
ch for PeerStatus, but since there's no actual peer in the
> attack, I don't seem to get an event from AMI.
>
> Any ideas?
>
There is an AMI security class that you can use to monitor the AMI security
events.
See manager.conf.sample
Richard
--
__
> There are certain versions of the Linux kernel that have no support
> under the older version of ESXI. We started having issues under our
> ESXI v4 setup with RH Enterprise and vmware's response was, "It's
> not supported"
"not supported" and "does not work" are not the same thing. ESXI
> The version is licensed and the customer does not want to invest on new
> hardware/software at the moment. If the ESXI version is too old I need
> to give them definitive proof that the segfaults are caused by that but
> since the old elastix has been running there for years they do not
sion 13?
>
> Any advise would be welcome.
>
The SetMusicOnHold application was deprecated in v1.6 and removed in v13.
Use
Set(CHANNEL(musicclass)=class) instead to set the music class on the
channel.
The change was documented in the UPGRADE.txt files.
Richard
--
___
>
> Related, Why can we have multiple Hangup handlers but not Pre-Dial
> handlers?
>
* There is only one dial to execute the called channel pre-dial handler
while there are many opportunities to specify hangup handlers.
* How do you think you could associate different pre-dial ha
an
> incompatible "codec" on both legs so it shouldn't switch to direct
> media.
>
> Has anyone else seen this issue?
>
This is an old issue. One of the latest issues is:
https://issues.asterisk.org/jira/browse/ASTERISK-25166
Richard
--
___
>
You are declaring an extension line with a pattern but the pattern only has
literal characters so it really isn't a pattern. It takes more CPU to match
than the non-pattern form and is more likely an error.
Richard
[1] ht
> It was only when I ran AsteriskLint over my dialplan that I noticed this:
>
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Application_Set
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Function_SET
>
> Hmmm, they both seem to do the same thing. Or don't they?
In some
ked anywhere a function
can be invoked and not just in dialplan.
Richard
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surrogate channels are replacement channels for masquerades to swap with
your target channel. They are created to die after a masquerade has
substituted it
for the target channel. If you are seeing them in dialplan then just let
them die
without doing anything else with them.
> Use menuselect's command line (--enable and --disable).
Great idea! How would you recommend generating the set of --enable and
--disable options that differ from the default from a build that was done?
--
_
-- Bandwidth and
> Of course, you might run into problems if the later release introduces new
> options (or deprecates old ones) which then aren't going to be in your
> makeopts file
That's my question: how do I reflect the changes that I made to the
defaults in a way that's not dependent on the exact set of
I'd like to be able to save the choices made in menuselect in a way
that they can be tracked in a CM system and applied to a later release
of Asterisk using an automated tool like Ansible. What's the best
way to do that?
--
_
ead of the ** and that works fine.
>
> Is there anyway to get the ** to work? I also am using a polycom phone if
> that affects things. I'm using asterisk 13.15.0
>
A ** extension should work just fine. I expect it is the dialplan in the
I had three crashes this morning on a divide-by-zero, for example at
abstract_jb.c:1008 in 14.3.0.
Does this ring any bell to anybody?
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Check out the
> The feed function in slinfactory explicitly does not allow frames
> without a data payload to be added to the queue. It would have prevented
> this crash.
Ah, so the fix should really be there, righty?
> I think the underlying issue is that the data pointer is not NULL when
> it sanely should
> All patches need to go into JIRA with a license agreement to be
> accepted.
Understood, but I was using it as an illustration. Note, however, that,
from a legal perspective, a patch such as this has no protectable IP (you
can't copyright the only way of doing something) and the GNU projects
Another crash with a packet:
$10 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0x12c62170, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 324, offset = 64,
src = 0x2ad290064a08 "siren14tolin32/speex", data = {ptr = 0x80893318,
> I would say this is a bug in func_speex and not in codec_siren14. This
> is because the datalen is zero.
Ah! So, like?
*** func_speex.c.orig 2017-02-13 15:00:19.0 -0500
--- func_speex.c2017-04-06 11:16:03.0 -0400
***
*** 185,189
}
!
I'm seeing Asterisk crashes with the following frame at func_speex.c:188:
(gdb) p *frame
$6 = {frametype = AST_FRAME_VOICE, subclass = {integer = 0,
format = 0xe2f9e20, frame_ending = 0}, datalen = 0, samples = 640,
mallocd = 1, mallocd_hdr_len = 232, offset = 64,
src = 0x2ac07413e7f8
context pjsip
config option.
The channel is reused to process each message in dialplan. It is invalid
for dialplan to
do anything with media on that channel and VoiceMail definitely falls into
that category.
Richard
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_MessageSend
ht
I recently upgraded to Asterisk 14.3.0. When playing a SIP file to a
G722 SIP channel (via chan_sip), I get a crash with the following
traceback. This is reproducable:
#0 0x0036fdc30265 in raise () from /lib64/libc.so.6
#1 0x0036fdc31d10 in abort () from /lib64/libc.so.6
#2
ConfBridge
or
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_ConfBridge
config show help app_confbridge user_profile template
or
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Configuration_app_confbridge
Richard
--
_
> I can't speak for the MRCP guys, but from a difference perspective,
> swapping MRCP from Asterisk 13 to Asterisk 14 shouldn't be too
> difficult. Most of the changes between the two shouldn't affect most
> people's use cases, including projects such as MRCP. I'd definitely
> check with their
When I look at the lastest UniMRCP manual, they only mention as high as
Asterisk 13. Does anybody know if I need to do anything to allow it
to work on Asterisk 14 and, if so, what that is?
--
_
-- Bandwidth and Colocation
quot;http://sIte.com:80/api/v1/
> calls?apiKey=UABVAEI=3")}
> executes and get answer from the server [{"RequestedCount":0,"
> MissedCount":7,"Total":7}]
>
The Set isn't being executed by the ExecIf. However the ${} substitution
containing
the
n Asterisk server in production with
> DEBUG_THREADS enabled ?
>
No, you should not leave DEBUG_THREADS on in a normal production
environment. Only enable it when you
are actually hunting for a deadlock. DEBUG_THREADS causes a noticeable
drop in performance.
Richard
--
>
Set the ATTENDED_TRANSFER_COMPLETE_SOUND channel variable to the sound
file to play on a transfer.
Richard
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Check out the new Asterisk community forum at: h
a pre-bridge handler (or any handler for that matter) is a
bad thing to
do and definitely falls into the "undefined behavior" category.
Richard
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Check out the n
uration and its
> interval?
>
>From the CLI do:
core show application SendDTMF
Right there in the documentation it says you can specify the intra-tone
time and the DTMF duration.
You can also look on wiki.asterisk.org for the SendDTMF
on should
be able to use all of the current libpri features.
Only DAHDI would really care about the kernel version and I cannot say if
that kernel is
supported with the latest DAHDI. The easiest way would be to try compiling
it.
Richard
--
__
bout in this scenario I describe (four contexts involved)?
>
In your case, the h exten is run by SIP/origin and Local/dest_ext_num;2.
These
channels executed dialplan when the call was originally placed. The h
exten runs
on the respectiv
>>
>
> Correcting myself, make uninstall seems to be what I was after for
> Asterisk itself.
> I'm still searching for the equivalent make target for pjproject.
>
pjproject has a make uninstall target as well.
Since v13.8, Asterisk has a --with-pjproject-bundled option [1].
On Sun, Nov 27, 2016 at 11:13 AM, Jonathan H <lardconce...@gmail.com> wrote:
> Thanks, Richard - your code does indeed work reliably 100% of the
> time, and thank you for that explanation.
>
> I do think the docs at
> https://wiki.asterisk.org/wiki/display/AST/Asterisk+14+Fu
sing variable
inheritance.
[svtest1]
exten = s,1,NoOp()
same = n,Answer()
same = n,Set(__MY_CALLER=${CHANNEL(name)})
same = n,Dial(Local/s@svtest2,,g)
same = n,NoOp(Returned SHARED(sharedVar) = '${SHARED(sharedVar)'}
same = n,Hangup()
[svtest2]
exten = s,1,NoOp()
and Local/s@dial-dest;1
When Local/s@dial-dest;2 executes Answer it will allow Local/s@dial-test;1
and ;2 to
optimize out because both ends are in a bridge. Thus the H dial option
will disappear from
the channel chain.
Richard
--
__
tting but I don't see an equivalent in pjsip.conf
>
> Do I need to use setvar to set CHANNEL(parkinglot) on my endpoint to do
> this now?
>
Yes. In your type=endpoint section you specify
set_var=CHANNEL(parkinlot)=mylot
Richard
--
_
On Sat, Sep 10, 2016 at 5:18 AM, Jonas Kellens <jonas.kell...@telenet.be>
wrote:
> On 10-09-16 09:42, Jonas Kellens wrote:
>
>
> On 10-09-16 00:50, Richard Mudgett wrote:
>
>
>
> On Fri, Sep 9, 2016 at 5:37 PM, Jonas Kellens <jonas.kell...@telenet.be>
&
g
> information about queues (I don't see this message on any other command).
>
That message is a result of trying to build a string where the buffer is too
small to contain it. I would expect that there is a truncated string in the
'queue show' output.
You haven't stated which Asteri
t is ringback with a recording interspersed at intervals,
you can create a
music-on-hold class and have the caller hear that instead.
Richard
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Join the Asterisk Comm
finds priority 3 of the generic series.
3, third in generic series
There is no priority 4 so the call is hung up.
Richard
[1]
https://wiki.asterisk.org/wiki/display/AST/Contexts%2C+Extensions%2C+and+Priorities
[2] https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching
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pjsip.
Also you should look here for more information:
http://blogs.asterisk.org/2016/07/13/asterisk-task-processor-queue-size-warnings/
Richard
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