the MagicJack will hear everything. If I change it
to 2, nothing is heard from that point on.
Chris
On 6/1/20 8:43 AM, Chris Dos wrote:
> I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and
> converted form SIP to PJSIP using the python script as a start and then
I upgraded our Asterisk 13 LTS to Asterisk 16 Certified 16.8 yesterday and
converted form SIP to PJSIP using the python script as a start and then
mofiying from there. I ran into an issue when testing that incoming calls
from MagicJack would go silent after about 10 seconds. This happened while
Hi,
https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981
The phone does work, you do need to TFTP the configuration files to the
phone though. Doesn't look like custom firmware is required.
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Chris.
On Fri, Apr 12, 2019 at 3:29 PM Antony Stone <
antony
,
Chris
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the same is true for much of
Europe.
Kind regards,
Chris
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+1
spending money to get that many fxs ports is going to negate any savings of
reusing analog phones instead of buying ip phones
1000 analog ports sounds like hell and if it was me I would be embarrassed
to have a setup like that tied to my name if I was a consultant etc.
Someone will come in
Has anyone done any integration of USB, etc. panic buttons and Asterisk?
The basic idea would be to have a USB based panic button[1] along with a
bit of code which would trigger a group SMS or perhaps a pre-recorded call
to a group.
Kind regards,
Chris
[1]http://www.amazon.com/StealthSwitch
=friend
dtmfmode=rfc2833
host=my_did_here.mobilet100.sipclient.org
context=vmobile-out
defaultuser=my_did_here
fromuser=my_did_here
trustrpid=yes
sendrpid=no
secret=super_secret_password
disallow=all
allow=ulaw,g726,gsm
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Chris
which seems to work for
Sipura/Linksys/Cisco phones, though most of my new deployments are exclusively
Snom.
Kind regards,
Chris
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people would
rather not deal with either group.
3) Lack of effective protection - both technical and regulatory -
against SIP-to-SIP misuse (not just fraud, but unsolicited callers, etc.)
Kind regards,
Chris
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regards,
Chris
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:
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Scott,
Somehow never noticed this setting before, I have tested it and it really
works great for the forwarding to cell situation we described. Thanks alot
for pointing me in the right direction.
chris
On Fri, Dec 19, 2014 at 9:39 AM, Scott Griepentrog sgriepent...@digium.com
wrote:
The main
as long
as it is multitenant or scalable beyond single server.
TIA,
chris
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That's the best analogy I've heard in favour of open development for a
long time, and something that non-techs can understand.
I thank you sir :-)
Kind regards,
Chris
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that
you'll want to consider, and the above won't work if you're using sipcli
legitimately anywhere on your network)
Kind regards,
Chris
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On 2/10/14 6:52 pm, motty cruz wrote:
Hello, our VoIP send us caller ID +1(area)(number) for instance
+16024224334 is there a way to strip +1 out of caller ID?
${CALLERID(num):1} should do what you're after (or :2 if you need to
strip the + as well)
Kind regards,
Chris
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files. They were designed to do it. Why try to shoehorn a database into
doing something it wasn't designed to do (and isn't particularly good at
doing)?
Kind regards,
Chris
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to share feel
free to reply on or off-list.
Thanks!
chris
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,
Chris
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want to
broadcast 1000+ simultaneous calls. Perhaps I'm just not being
imaginative enough... :-)
Kind regards,
Chris
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and 840 drives.
Kind regards,
Chris
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speech-to-text engines has been one
of poor accuracy at best.
If you need messages-to-text, generally best to use a virtual PA company or
similar - at least in my experience.
Kind regards,
Chris
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, manipulating
them, then displaying the results will likely exceed that of a DB.
Unless you only want a recent call log, you really want to do this in a
database.
Kind regards,
Chris
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, but if you're phoning people who've not
consented to sell them double glazing, then you aren't doing your
client's reputation any favours and you're going to mightily annoy those
you're calling.
Kind regards,
Chris
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to
redo all the AEL macros from 1.4 does not fill me with any enthusiasm to
update those boxes.
The switch to Gosub() does not seem to be an easy drop-in replacement
for Macro().
Kind regards,
Chris
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to a storage
device at regular intervals: make sure you use lsof or similar to check the
recordings aren't actually open by asterisk at the time, otherwise interesting
things will happen.
Kind regards,
Chris
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11 would be gratefully appreciated)
Kind regards,
Chris
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threaded readers.
That is all :-)
Kind regards,
Chris
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the
manager interface to a dummy extension and make sure it completes
successfully.
FWIW, we tend to use pacemaker with heartbeat rather than corosync, but
both perform a pretty similar function.
Kind regards,
Chris
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should speak to a qualified legal professional to allay your
concerns. This list has such an international audience that what's
perfectly acceptable in one jurisdiction might land you in hot water in
another.
Kind regards,
Chris
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probably safest to send RTP traffic through Asterisk
regardless, otherwise you're potentially opening up a support nightmare
for yourself.
Kind regards,
Chris
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' service on your asterisk
platform with all the normal asterisk VM benefits such as email
delivery, etc.
You can then of course detect when those mobiles 'divert' to voicemail
(since it's now on your system), and kick them out of the queue at that
point.
Kind regards,
Chris
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devices in question and your mobile
operator will allow their voicemail service to be completely disabled.
Kind regards,
Chris
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recording to a ramdisk first, then periodically write
out completed files to HDD at your leisure (e.g. during slack periods)?
Or, given the relatively low cost of 250GB SSDs these days, swap out the
spinning disc for an SSD.
Kind regards,
Chris
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of your users travel internationally, I'd be
wary of imposing countrywide IP blocks, especially in this era of IP
shortage where IP space is being traded on the open market and GeoIP
databases may not always keep up to date.
Kind regards,
Chris
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on
other variants such as Australian or American.
Kind regards,
Chris
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and/or cache
settings - this might be something worth asking about on the respective MySQL
discussion groups as well as here.
Kind regards,
Chris
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Via Epia MiniITX
systems which don't have full i686 instruction set support).
The best Linux distro is usually the one you're most familiar with -
that way, if/when something goes wrong, you stand a reasonable chance of
being able to fix it.
Kind regards,
Chris
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is never a bad
thing, and ideally on SSDs or RAM storage if you can. Spinning disks are
often the bottleneck with large data sets.
Kind regards,
Chris
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://directory.pioneer.world. There you
can leave an emergency message for the on-call technician.
Otherwise I will respond as soon as I can.
Thanks,
Chris Douglas
Technical Services Manager
Pioneer Balloon Company
tel. 316-688-8648 fax. 316-691-6901
Note: This is an automated response to your
attacks...
Kind regards,
Chris
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be possible for the second line to catch some false matches, but I
have not seen any issues with our system thus far.
Kind Regards,
Chris
PS. Feel free to comment on what is wrong with this and be sure to include
the right way to do
everyone else's experiences have been like, and why
95% or better of the SIP attacks on my PBX are destined for Paltel mobile
subscribers.
Perhaps the termination payout on those numbers is particularly good,
and/or regulation/investigation into abuse isn't so good?
Kind regards,
Chris
*/extensions.conf
Which will include the file extensions.conf from each subdirectory. Very
handy if you have a structure like this:
/etc/asterisk/client1
/etc/asterisk/client2
etc.
Kind regards,
Chris
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was misunderstood, please let us know and provide
clarification. Thanks again!
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Digium, Inc. | Network and Computer Systems Administrator
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
main: +1 256 428 6000 | fax: +1 256 864 0464
of paying a third party with a suitable API
x per message to deliver them on your behalf.
Kind regards,
Chris
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?
Have a chat with your usual network equipment supplier for midspan PoE
units. Phihong make some, and those I've used seem to have been pretty
reliable. There are no doubt many other suppliers of such things.
Kind regards,
Chris
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not had any failures since (out of a few dozen).
The management interface isn't great, especially if you're used to the
command line goodness of an HP or Cisco unit, but provided you aren't
fiddling with it too often, you'll manage.
Kind regards,
Chris
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ulaw
On Wed, Jul 10, 2013 at 7:40 AM, basteon bast...@gmail.com wrote:
Hi,
What codec do you use with yours subscribers?
On 9 July 2013 23:45, Chris Gentle gent...@gmail.com wrote:
Is there any way I can improve the audio quality in a confbridge in
Asterisk 11? I've changed
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth mr...@imminc.com wrote:
The sampling frequency for u-law is 8,000 Hz. You can't produce a recording
with higher quality than the source, so you'd have to switch to a wideband
codec
to improve the conferences and recordings [1] [2].
OK, thanks
OK, thanks for the advice. No, there's no filter so I'll look into that.
On Wed, Jul 10, 2013 at 3:02 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 07/10/2013 06:46 PM, Chris Gentle wrote:
[snip]
and then others can connect via SIP. For some reason, when the
speaker says
missing something?
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Not to mention that installing Pacemaker/Heartbeat/Corosync or your
other HA solution of preference isn't particularly difficult, and is
agreeably free.
Kind regards,
Chris
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mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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On 06/13/13 14:30, Chris Gentle wrote:
Google the number and you can probably find other complaints and
possibly who it is. Not that it will matter, there's nothing you can
do but block it.
My approach to call filtering is:
Deny All
Allow Some
I have a whitelist
On 6/6/13 4:53 am, Gopalakrishnan N wrote:
Any other HA applications available or the lsyncd with pacemaker is good?
I generally use Pacemaker with Heartbeat, which seems to work pretty well.
Kind regards,
Chris
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(/home/asterisk/asterisk-ices-1000.xml)
;exten = 1000_ices,n,Hangup()
;}}}
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results.
If both (1) and (2) are successful, than there's some impact that the
Ices application is having on the Local channel that is messing up the
reference counting inside the ConfBridge. Otherwise, it's an error in
ConfBridge.
So what do you think? Should I file a bug?
--
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is using codec gsm. I am having one way audio
and getting below mentioned warning. Asterisk version is 1.8.11.0
Isn't g723.1 considered pretty poor quality these days? Can't you set
voipswitch to use something apart from that?
Kind regards,
Chris
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: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
snip
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New
on...
you could enable sip tracing to get more information.
maybe you should change the 'allowguest' option in sip.conf..?
regards,
yves
Am 31.05.2013 23:57, schrieb Chris Gentle:
OK, I need a bit of help here. I'm configuring a new Asterisk 11
system and I accidentally let my firewall rules
target audience. Recorded calls
always do.
Kind regards,
Chris
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that in your distro's startup scripts.
On Gentoo, you'd do something like /etc/init.d/logrotate start to
start it now, and rc-update add logrotate default to add it to your
default runlevel.
Difficult to advise further without knowing the distro in question.
Kind regards,
Chris
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at the moment (not just for asterisk stuff, but in general).
Kind regards,
Chris
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SIP connectivity and also their PSTN
termination.
Kind regards,
Chris
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kind of interrupt hammering going on here with
my particular hardware. Even before the audio completely fell apart I
could hear some little pops that sounded like the interrupts were not
being serviced fast enough.
On Mon, May 6, 2013 at 8:31 PM, Chris Gentle gent...@gmail.com wrote:
OK
ideas what I'm missing here? Is there a better way to do this?
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said, the OP might be best advised to request (paid)
assistance with the project on the [asterisk-biz] list.
Kind regards,
Chris
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if they do,
might not understand how to read SMS messages on their phone.
Probably would work okay for certain establishments, but I'd be wary of
exclusively using SMS in a medical context, given the likely patient
demographic.
Kind regards,
Chris
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they really are
talking to the patient (perhaps by asking their DOB or similar).
It may be possible to outsource something like this to a Virtual PA
service or similar.
Kind regards,
Chris
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emergency traffic. The county seems interested in exploring the
possibility.
So I'm wondering if anyone else has attempted this.
Kind Regards,
Chris
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not follow SIP RFC
http://en.wikipedia.org/wiki/Next_Generation_9-1-1. That is not saying
your county is not using standard SIP for E911, it just wouldn't be
considered NG911.
--
*From:* Chris Nighswonger
*Sent:* Fri 4/19/2013 11:41 AM
*To:* Asterisk Users Mailing List
carry cell phones, the soft phones being mainly remote
extensions.
So it sounds like it is at least worth pursuing.
Kind Regards,
Chris
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On 12/4/13 4:38 pm, Nick Khamis wrote:
We were looking more into the lines of a
scalable multi server router like a cisco 3745.
Perhaps it might help to tell the list just how many concurrent calls
you're looking to transcode?
Kind regards,
Chris
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it to 127 and it has worked ever since.
Yep, that's the same article I found and mentioned in original post.
Thanks for posting the link. While the article was written for
Asterisk 1.8 and Jabber, the same setting works in the xmpp.conf file.
Very useful stuff.
--
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this helps somebody else.
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Freepbx is over writing the conf files ?
On Mar 15, 2013 8:11 AM, Luis H. Forchesatto luisforchesa...@gmail.com
wrote:
Greetings.
I'm running asterisk here (elastix) and I have a few extensions configured
in it. I have here two different callgroup/pickgroup where the extensions
are
for another day or so. Has anyone else noticed this?
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something
that meets the above. I suspect many of us would find such a product or
application useful from time to time.
Kind regards,
Chris
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Awesome, thanks. I'll give it a try.
On Mar 11, 2013 4:56 PM, Joshua Colp jc...@digium.com wrote:
Chris Gentle wrote:
I'm currently running Asterisk 11.2.1 and I've noticed that when
asterisk has been up for a while (usually about a day), outgoing calls
through GoogleVoice fail to complete
don't know what pricing is like in your local currency of course. I
believe Yealink do also have a fairly reasonable remote provisioning
system, but unlike the Snom system, I can't claim to have used it in anger.
Kind regards,
Chris
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.
Also worth making sure you have retry=0 in your queue config.
Kind regards,
Chris
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Chris
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handset.
Kind regards,
Chris
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Chris
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about analogue
phones at all.
A friend did this down the length of a heritage railway as they already
had cable running the length of their tracks, and I believe it was
fairly successful.
Kind regards,
Chris
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for
you to knock up something similar for your property.
Kind regards,
Chris
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. And it's always worth disabling any SIP ALG present
in the router - they seem to do nothing but break things.
(as a random aside, has anyone *ever* come across a scenario where a SIP
ALG in a consumer router has actually helped?)
Kind regards,
Chris
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the blue pair.
So if all you need is a single line pair, you should be able to just
wire up the blue pair to the centre pins on your RJ11 connector.
Alternatively, you can cheat, and just use an RJ11 - RJ11 cable - these
usually fit just fine into an RJ45 socket.
Kind regards,
Chris
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Chris
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voicemail
and recordings, and to make those available across a multi-server
environment.
Let the discussions begin.
Kind regards,
Chris
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of Monitor():
http://www.voip-info.org/wiki/view/Asterisk+cmd+Mixmonitor
One of its options allows you to execute a command at the end of
recording, which you can then use to call a script to handle your
recordings however you wish.
Kind regards,
Chris
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communication.
Happy new year to you all.
On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani mi...@enterux.in wrote:
+1 here.
On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
On Wed, Jan 9, 2013 at 2:02 PM, Doug
this precisely the raison d'être for [asterisk-biz]?
Kind regards,
Chris
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of the word
business).
Christian Savinovich
VoIP Telephony Consultant
646-982-3572
Original Message
Subject: Re: [asterisk-users] DIDForSale spam
From: Chris Bagnall aster...@lists.minotaur.cc
Date: Thu, January 10, 2013 5:17 pm
To: Asterisk Users Mailing List - Non
I've seen this implemented on polycom phones where a secondary extension is
on the phone that is setup to auto answer and they have something on the
PBX side that is configured to call some or all of the secondary extensions
On Jan 10, 2013 8:28 PM, Carlos Alvarez car...@televolve.com wrote:
Lol yes it was all local on a gigE network even :) I also didnt say it
was the most elegant solution but it seemed to work well with them
they even had grouped it into extensions and I'm sure you could even
write some logic to make sure the calls are local
On Thu, Jan 10, 2013 at 9:31 PM, Carlos
, when I asked in March where he got my email he said:
Hi Chris,
We got your contact from the Internet. Let me know the good time to
talk about this in detail.
Thank you,
-Rohit Dhaka
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asterisk 11 packages for the Pi? I ended up
compiling it myself this weekend. Took a while.
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