files
available in all the possible native formats. Then Asterisk can use the
appropriate
one for the channel without transcoding.
On my systems I have MoH and sounds installed in wav, ulaw, alaw, gsm and g729.
They will also sound better than transcoding from the gsm versions.
Cheers
T
xample) is not used to select or match the inbound SIP
peer.
When the call comes in from sipgate, it probably doesn't have a fromuser.
The fromuser can be used to select the peer based on matching the [string]
that names the peer.
Otherwise, when Asterisk is looking for a matching peer sect
l(SIP/officephone,120,m)
>
> [secondline]
> exten => 22,1,Dial(SIP/livingroomphone,120,m)
>
> [thirdline]
> exten => 33,1,Dial(SIP/bedroomphone,120,m)
But because you have all three of your trunk peers pointing to the
same context, you don't necess
o send to an unreachable peer), that may set
DIALSTATUS without setting HANGUPCAUSE.
So HANGUPCAUSE should be considered as extra detail, rather than a replacement
or alternative to DIALSTATUS.
Cheers
Tony
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oes anyone have any clues why there would be a difference
in PRI behaviour between 32-bit and 64-bit builds? Has anyone else run into
anything similar?
Cheers
Tony
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In article ,
Matt Fredrickson wrote:
> On Tue, Apr 3, 2018 at 5:44 AM, Tony Mountifield wrote:
> > I have some more investigation to do on this, but I wanted to see if anyone
> > here had any insight into the issue I've run into.
> >
> > The hardware is a HP DL360
degrade to half-duplex trying to talk
to full-duplex, resulting in lots of collisions and packet loss when there
is any kind of significant traffic.
Your description would be consistent with the firewall introducing lots of
LAN collisions when busy, in the central gigabit switch, even if the VoI
In article ,
Matt Fredrickson wrote:
> On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote:
> > In article
> > ,
> > Matt Fredrickson wrote:
> >> That does seem quite odd. If I remember right, those messages would
> >> come up if it looked like the o
when dahdi_cfg couldn't find libtonezone).
Would there be any subtle issues with the 64-bit libraries being loaded
from /usr/lib instead of /usr/lib64?
Should Asterisk and DAHDI builds also be updated to use /usr/lib64 when
building on a 64-bit OS? Or the build instru
In article ,
Tony Mountifield wrote:
> In article
> ,
> Matt Fredrickson wrote:
> > On Tue, Apr 3, 2018 at 4:38 PM, Tony Mountifield wrote:
> > > In article
> > > ,
> > > Matt Fredrickson wrote:
> > >> That does seem quite odd. If I rem
In article <20180404133024.kpidrkuiyjoqd...@xorcom.com>,
Tzafrir Cohen wrote:
> On Wed, Apr 04, 2018 at 11:28:33AM +0000, Tony Mountifield wrote:
> > In article
> > ,
> > Richard Mudgett wrote:
> > >
> > > The libpri makefile doesn't install
nt ast_do_masquerade(struct ast_channel
} exchange;
struct ast_channel *clonechan, *chans[2];
struct ast_channel *bridged;
+#ifdef I_THINK_THIS_IS_WRONG /* Tony Mountifield, 2018-03-29. Removing this
code fixes lost CDRs with masquerade */
struct ast_cdr *cdr;
+#endi
In article <7d8dc02f-0fce-4d47-72d9-604994c33...@palosanto.com>,
Alex VillacÃÂs Lasso wrote:
> El 29/05/18 a las 05:24, Tony Mountifield escribió:
> > In article <3a005ff6-19a4-215b-4751-bee616ec7...@palosanto.com>,
> > Alex VillacÃÂÃÂs Lasso wrote:
> >&g
ay to tell AMI that I don't want it to log login attempts - or,
> to put it another way, is there any way to tell the logger module to ignore
> AMI?
Look in /etc/asterisk/manager.conf for the option "displayconnects = yes/no".
It can be set globally in [general] or indiv
binary, recompile, then compare the first binary with
the recompiled one? At the simplest level use "cmp -l". Or maybe convert
each binary to a hexdump with "hexdump -C", and then use diff or vimdiff
to compare them.
Cheers
Tony
-
Dial() to propagate the answer, busy or other failure from
the destination channel back to the originating channel.
Is it possible that the setup part of the call (between initiation and answer)
is recorded in a separate CDR?
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.
ording on Asterisk 13?
>
> (This obviously is fatal anyway as I got lots of phones on which I want to
> playback recordings and editing /etc/hosts for each phone is impossible if
> two phones want to listen to different recordings at the same time-
> /etc/hosts can only contain one &
languages/libraries I should be considering?
Thanks for any advice!
Cheers
Tony
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--
_
-- Bandwidth
In article <301a2e78-d490-3805-e30f-41b668aac...@sysnux.pf>,
Jean-Denis Girard wrote:
>
> Hi Tony,
>
> Le 20/07/2019 à 06:29, Tony Mountifield a écrit :
> > Are there any other languages/libraries I should be considering?
>
> Same here, after years of AGI /
In article ,
Jean-Denis Girard wrote:
> Le 20/07/2019 à 12:21, Tony Mountifield a écrit :
> > What is the bug with channel variables? Do you have a fix for it?
>
> Channels variables caused an error, my fix is in aioswagger11/client.py
> (line 80)Â :
>
Who is the male voice artist who recorded the en_GB sounds for Asterisk?
Would be useful to know in case of the need to get additional matching
sounds recorded.
Cheers
Tony
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to
handle "qualify".
So in your [trunkinbound] context, just add a line like this:
exten => s,1,Hangup
And leave everything else in that context unchanged.
Cheers
Tony
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Play: t...@
oad. Nevertheless, we will try what you just posted.
Even if you put "exit 0" at the top of the script, the perl interpreter will
still need to compile the whole script (and any modules it uses) before it
executes the "exit 0".
Try commenting out or removing the rest of the
ing to
Asterisk or from Asterisk.
Do you have internal_timing set in asterisk.conf?
What timing module are you using?
Does it always happen, or just sometimes?
Cheers
Tony
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2. Macro(record)
[extensions.conf:881]
3. Set(CALLERID(num)=044111)
[extensions.conf:882]
-= 2 extensions (5 priorities) in 1 context. =-
Notice that the "n" converted to "4&quo
t need to speficy the __ when reading the variable, just use ${PAI}
as before.
See https://wiki.asterisk.org/wiki/display/AST/Variable+Inheritance
Cheers
Tony
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--
_
line you quoted:
same => n,NoOp(PAI=${PAI})
Then turn on verbose logging and try the call. Look at the logged
NoOp line and see if it contains just the 'John' or the whole value
'"John Doe" '
If it contains the whole value, then the problem is in the AGI library
m I missing something?
I agree with you that it is strange the two logging types are different.
But someone with a different opinion than yours might well say "Why did
they decide to omit the line number and function from the file logging?
It's very useful information!"
The be
In article <88f96e46-e6bb-a7ef-bebb-5588ef6cd...@gmx.ch>,
Fourhundred Thecat <400the...@gmx.ch> wrote:
> > On 2020-06-02 17:48, Tony Mountifield wrote:
> > In article <94191802-6c9c-bdab-615b-001786a2a...@gmx.ch>,
> > Fourhundred Thecat <400the...@gmx.
ip entry, for OPTIONS to return a 200 instead of 404.
It doesn't matter what the 's' extension does, so it can just call Hangup.
Cheers
Tony
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__
- you will find it much more reliable.
Cheers
Tony
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a situation that doesn't make much sense). It looks above like you
did "chmod 6644 /home/silentm/bin/conf-background.agi". Try again with
0755 instead of 6644.
Cheers
Tony
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ad() before the
> Playback is finished?
Try this:
exten => 100,1,Answer()
exten => 100,n,Read(OPTION,LONG-MESSAGE,2)
Cheers
Tony
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ue executing the AGI program or to issue a
> HAGNGUP explicitly.
When you have returned from the Dial command, check the DIALSTATUS channel
variable. If the call was picked up, ${DIALSTATUS} will contain ANSWER.
Cheers
Tony
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Work: [EMAIL
gt; Pete
> > _______
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/
thing
wrong with IE compatibility.
Does anyone know what it would take to make the GUI compatible with IE
as well as FF?
Cheers
Tony
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__
asterisk-gui/branches/asterisknow and that trunk has lagged behind. Is that
correct?
Cheers
Tony
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In article <[EMAIL PROTECTED]>,
Roderick A. Anderson <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > When I bring up the Asterisk GUI in AsteriskNOW, using IE7, it displays
> > a message at the top "Your browser is not supported by this version of
> &
e, and the interrupt counts
on there should be going up at approximately 1024 per second.
What kernel version are you using? (uname -a)
Cheers
Tony
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___
In article <[EMAIL PROTECTED]>,
Jerry Geis <[EMAIL PROTECTED]> wrote:
> > On Thu, Apr 03, 2008 at 01:45:09PM +, Tony Mountifield wrote:
> >
> > >/ Jerry, the first thing to check is "cat /proc/interrupts" and see if
> > >there
> >
6.9 with no problems, and again on 2.6.18-53.1.6.el5
Unless it's a 64-bit issue - I've only ever used 32-bit.
Cheers
Tony
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_
rtc errors in dmesg either.
>
> I have tried 2.6.24.4 and same situation. rtc just has 1 in
> /proc/interrupts.
>
> What now?
Is this system used for other things, or could you try installing a
32-bit (i386) version of CentOS and trying again? 64-bit vs 32-bit
is the only think I
In article <[EMAIL PROTECTED]>,
Tony Mountifield <[EMAIL PROTECTED]> wrote:
>
> I have just installed ztdummy on a new system running 2.6.18-53.1.6.el5,
> and it is incrementing fine. I didn't realise there was a newer kernel out;
> I'll have to update and
watch /proc/interrupts.
5. Also do "service asterisk start".
6. When the system is booted up in future, zaptel and asterisk should
automatically start.
Cheers
Tony
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n alternative to AsteriskNOW, unless they discard their existing
hardware and buy yours.
It bothers me that a thread I started about a technical issue has been
used as a pretext for a commercial promotion that doesn't really address
the issues at hand.
Cheers
Tony
--
Tony Mountifield
Wor
calling or the called party on his current call.
If he is the calling party, he will execute 'h' when he clears, but
if he is the called party, he won't be in the dialplan to execute 'h',
so we need some other way to invoke the ringback (step 5).
Thoughts?
Che
eeps on trying until it runs out of retries.
Cheers
Tony
> On Fri, 2008-04-04 at 10:35 +, Tony Mountifield wrote:
> > Has anyone here implemented "Ring back when free" in Asterisk?
> >
> > The way it works in the UK is as follows:
> >
> > 1. A call
; This keeps the real time clock ticking and playing wave files.
>
> Thanks to all!
Jerry, glad you got it working. acpi=off did cross my mind, but I don't
really understand what it does, nor why it would help!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http
In article <[EMAIL PROTECTED]>,
Michiel van Baak <[EMAIL PROTECTED]> wrote:
> On 10:35, Fri 04 Apr 08, Tony Mountifield wrote:
> > Has anyone here implemented "Ring back when free" in Asterisk?
> >
> > The way it works in the UK is as follows:
> >
it's seldom used in the UK,
and I thought it meant something more like Call Waiting than Ringback When Free.
Cheers
Tony
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__
pplications" and look at what apps are
> included with Asterisk.
I do, frequently!
Cheers
Tony
> Tony Mountifield wrote:
> > Has anyone here implemented "Ring back when free" in Asterisk?
> >
> > The way it works in the UK is as follows:
> >
> >
right and it doesn't work, it's the card. If it
does work, then your problem is with the carrier.
Cheers
Tony
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___
-- Band
o turn on
SIP debugging at the Asterisk CLI> prompt to see the packets sent to/from
Asterisk.
Cheers
Tony
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-- Bandwidth
nt: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Supported: replaces
> Content-Length: 0
>
> Why is it doing so?
>
> On Thu, Apr 17, 2008 at 2:36 AM, Tony Mountifield <[EMAIL PROTECTED]>
> wrote:
>
> > In article <[EM
alplan won't fall through a gap.
The only way to reach a priority after a gap is to Goto it in some way.
In your example above, if you return from priority 20, it will look only
for 21, and if not found, it won't keep looking for higher priorities.
Cheers
Tony
--
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Work
leases to be x.y.z, and don't see the point in doing
an x.y.z.a just because a change is small.
Cheers
Tony
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___
-- Ban
s complete:
exten => 233,1,SetAccount(queue1)
exten => 233,2,Queue(queue1|rn)
exten => 233,3,NoOp(${QUEUESTATUS})
exten => 233,4,NoOp(${DIALSTATUS})
exten => h,1,NoOp(${QUEUESTATUS})
exten => h,2,NoOp(${DIALSTATUS})
Cheers
Tony
--
Tony
en addressed in SVN trunk by the addition of the
option F(context^exten^pri) - When the caller hangs up, transfer the
called party to the specified context and extension and continue execution.
However, it doesn't appear to be in the 1.6.0 branch, so won't appear in
a release until 1.6.1.
If
uot;. That should make it play out based on
internal zaptel timing instead of timing off the incoming stream, I think.
Cheers
Tony
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quot; and check that the RTC interrupts
are going up by 1024 per second. This is with ztdummy running.
What else is going on on this server? Does it have any virtual machines
on it? Does it have X Windows running? What does "top" show?
Cheers
Tony
--
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Work: [
n in the first place.
The only dependency it has is that the kernel must have been built with
CONFIG_RTC and not CONFIG_GENRTC.
> And then again, on kernels >= 2.6.22 you have hi-resolution timers which
> generally work better.
I have yet to experience these, but it sounds promis
n
number as '12127551200' with a TON of International. Dialling a UK number
01234 567890 would be sent as '1234567890' with a TON of National.
The second set of parameters sends all numbers as-is, but with a TON of
Unknown. This may allow the exchange to do normal interpretation
Does anyone know if the Digium PRI cards can be configured or modified
to have a high-impedance input on the RX pair? I would be interested in
this in order to build a bi-directional PRI audio sniffer using two
E1/T1 ports per trunk to be monitored.
Cheers
Tony
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ther Digium cards can be set to hi-Z if anyone knows.
Cheers
Tony
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; > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> Best Regards
> Rizwan Hisham
>
> -=-=-=-=-=-
> [Alternative: text/html]
> -=-=-=-=-=-
> -=-=-=-=-=-
&
nent-Client-Asterisk-Manager/
You would need to use the POE event-driven framework.
Or you could try Asterisk::Manager in asterisk-perl-0.10
I haven't tried either of them - most of my Manager stuff is in C.
Cheers
Tony
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Play:
necting
channels without dropping calls when one side hangs up.
I've been looking at VICIDIAL for the first time this week, and am at
last beginning to understand why it does things the way it does. :-)
It's a good source of ideas as well as something to use in its own right.
Cheers
Tony
he colors.
> Unless ofcourse I manually stop now, and safe_asterisk.
>
> What gives?
> I just want to have an auto restart, and CLI with colors!
Cheers
Tony
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, between
Voicetronix and Sangoma cards? Or are wanpipe and wanrouter generic
software components that both companies just happen to use?
Cheers
Tony
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - ht
Try adding "reinvite=no" to the sip.conf or users.conf definition for your
SIP service provider.
Cheers
Tony
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___
-
nvolved).
Putting "reinvite=no" in sip.conf for either endpoint (but best to do it
for the Service Provider endpoint) tells Asterisk never to optimise
itself out of the media path, even if "t" and "T" are not specified.
Cheers
Tony
> Cheers,
>
> Moe
>
ied
running all 120 channels at once!
Cheers
Tony
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as
nd the called party
goes to priority pri+1, so that you can do different things for each.
Cheers
Tony
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-- Bandwidth an
ng between two Asterisk boxes, so am
just after some confidence regarding the Avaya, or any gotchas to be
aware of.
Cheers
Tony
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e bridge properly), and
doesn't creak under the bloat of OpenH323 like the first two do.
I don't know whether Objective Systems have abandoned chan_ooh323 and
the ooh323c stack, but it would be great to see them moved from -addons
into the main Asterisk tree.
Cheers
Tony
--
T
the first two. It is clean
> and lightweight,
> uses the Asterisk RTP stack (and can therefore bridge
> properly), and
> doesn't creak under the bloat of OpenH323 like the
> first two do.
>
> I don't know whether Objective Systems have abandoned
> chan_ooh323 and
> t
re may well be other issues, but fix these first. Then if you still
get problems, include the verbose output from the log file.
Cheers
Tony
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___
e other party.
There have been one or two enhancements proposed in the past to allow
one channel to grab another and bridge to it, but I don't think such an
application has made it into official versions yet (1.4 or trunk).
Cheers
Tony
--
Tony Mountifield
Wo
7;
> == Spawn extension (na-miasto, TestTrakt, 1) exited non-zero on
> 'SIP/sempron-b2ae1918'
>
>
> any idea how to force Asterisk to push call via particular timeslot?
Not sure where you got the idea to use 1-2 as the channel number.
Just use Zap/1/517255333
just doing a standard "round to nearest" integer division, by adding
half the divisor to the dividend before dividing. Without that, you just
get "round down" instead.
Cheers
Tony
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ere
> an obvious way of seeing what's going on at the moment?
show channels
Cheers
Tony
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, but I see the message is still there in Trunk.
Cheers
Tony
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In article <[EMAIL PROTECTED]>,
Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> Tony Mountifield wrote:
> > When making or receiving a SIP call via my service provider, I get the
> > following message logged by Asterisk:
> >
> > Dec 11 15:13:37 NOTICE[7392]: r
ove to 1.4, any customisations or new
features I do create would still need to be ported to trunk before they
would have any chance of making it into Asterisk. This takes time, which
is always in short supply, and means that some cool features remain mine
only :-(
Cheers
Tony
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Tony Mountifie
e formats", before calling MeetMe. e.g.
exten => _X.,1,Set(MEETME_RECORDINGFORMAT=gsm)
exten => _X.,n,MeetMe(${EXTEN},dr)
Hope this helps!
Cheers
Tony
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__
and
2) How you are attempting to send the tones.
Please give more details, and hopefully we can help you.
Cheers
Tony
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___
-- Ba
of kernel
jiffies.
If that's not it, then I hope you discover the cause, and would be
interested to know what it is.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> On Tue, Feb 05, 2008 at 09:58:54AM +0000, Tony Mountifield wrote:
> > Specifically, what kernel version?
>
> As I understand from an IRC chat - CentOS 5 - 2.6.18-something.
Hmm, ok.
> > If you are using 2.6.9 (e.g. Centos 4
In article <[EMAIL PROTECTED]>,
Steve Edwards <[EMAIL PROTECTED]> wrote:
> On Tue, 5 Feb 2008, Tony Mountifield wrote:
>
> > Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
>
> >> So that point is mute.
>
> > "moot", not "mute".
stripped from the
number, and a ToN of "international" is attached. Else if the national
prefix matches, that is stripped and a ToN of "national" is used.
Otherwise the number is left unchanged an a ToN of "local" is used.
Hope this helps!
Cheers
Tony
--
Tony Mountif
(muting conference ${CONF})
exten => 5,n,MeetMeAdmin(${CONF},N)
exten => 5,n,Goto(${SAVEDEXTEN},1)
; allow user to press 6 to unmute all users
exten => 6,1,NoOp(unmuting conference ${CONF})
exten => 6,n,MeetMeAdmin(${CONF},n)
exten => 6,n,Goto(${SAVEDEXTEN},1)
Hope this helps!
Cheers
RI, Frame Relay, Linux, and network design. Based near
> Birmingham, AL. Now accepting clients worldwide.
>
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> asterisk-users mailing list
> To
I'd be interested in any stories of success
or failure.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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call recording (r option), and MeetmeAdmin
operations such as mute all and unmute all.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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ions :-)
Cheers
Tony
> MATT---
>
> On 3/4/08, Tony Mountifield <[EMAIL PROTECTED]> wrote:
> > Has anyone here done any work on clustering Meetme conferences over
> > multiple Asterisk boxes? The scenario I am thinking of is where there are
> > two or more boxes co
started up, and incrementing by 1
for each subsequent channel.
In Asterisk 1.4 or later, an optional system name can be defined in
asterisk.conf, and if defined, the unique ID becomes:
-.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
but precede it with a call to the Read()
application for the user to enter their conference number. This will put
it into a channel variable, e.g. ${CONF}, which you can then put in place
of the hard coded number.
Cheers
Tony
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Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.u
r a legacy system),
and also whether it is any different on later versions.
Thanks,
Tony
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Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
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In article ,
Paul Belanger wrote:
> On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield wrote:
> > I haven't been able to find the answer online, and am not currently
> > able to conduct an experiment to find the answer...
> >
> > I understand that in a SIP call w
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