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 i
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.
--
Chris.
On Fri, Apr 12, 2019 at 3:29 PM Antony Stone <
a
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
AIUI the same is true for much of
Europe.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community
+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 af
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
obile SIP user
[my_did_here]
type=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=al
ich seems to work for
Sipura/Linksys/Cisco phones, though most of my new deployments are exclusively
Snom.
Kind regards,
Chris
--
C.M. Bagnall
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation
e fraudsters and telemarketers, and most 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
--
This em
press.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
SCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introd
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
wrote:
> The main problem we are try
long
as it is multitenant or scalable beyond single server.
TIA,
chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
.
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
--
This email is made from 100% recycled
On 3/10/14 6:52 pm, Rainer Piper wrote:
the attacking server changed the destination Number at 18:53 CEST and
he is still blocked ... LOL
972597438354
It's pretty much an everyday occurrence for any internet-connected SIP
system these days...
Oct 3 19:46:20 server /sbin/kamailio[3977]
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
--
This ema
at storing
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
--
This email is made from 100
riences to share feel
free to reply on or off-list.
Thanks!
chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
t
performance under load remains to be seen.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for
you'd want to
broadcast 1000+ simultaneous calls. Perhaps I'm just not being
imaginative enough... :-)
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation P
isks with a mix of Samsung 830 and 840 drives.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a l
o-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
--
This email is made from 100% recycled ele
anipulating
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
--
This email is made from 100% recycled electrons
--
___
chedule, then fair enough, 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,
C
11 would be gratefully appreciated)
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
ings 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
--
This e
having 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
--
This email is made from 100% recycled
To headers.
Completely breaks threaded readers.
That is all :-)
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? J
up a call using 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
--
This email is made from 100
on, you
really 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
--
This email is made from 100
these (and no doubt other) scenarios
don't apply, it's probably safest to send RTP traffic through Asterisk
regardless, otherwise you're potentially opening up a support nightmare
for yourself.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
g
end-to-end control of the mobile devices in question and your mobile
operator will allow their voicemail service to be completely disabled.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwi
wn 'mobile voicemail' 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.
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
--
This email is made from 100% recycled
thless set of rules/limits.
Unless you know that none 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.
K
other variants such as Australian or American.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
th a busy database, you might want to look at your MySQL indexes and/or cache
settings - this might be something worth asking about on the respective MySQL
discussion groups as well as here.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
n old 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
--
This email i
B on a separate host 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
--
This email is made from 100% recycled electrons
--
__
://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
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
TICE.* .*: Failed to authenticate device .*\s?\\>.*
and that handles most of the hacking attempts I see on my system. I think
it may be possible for the second line to catch some false matches, but I
have not seen any issues with our syst
#x27;d be curious to know what 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
*/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
--
This email is made from 100% recycled electrons
ipermail/asterisk-announce/2007-July/85.html).
If your response was misunderstood, please let us know and provide
clarification. Thanks again!
--
Chris Hozian
Digium, Inc. | Network and Computer Systems Administrator
445 Jan Davis Drive NW - Huntsville,
is
usually more than the cost of paying a third party with a suitable API
per message to deliver them on your behalf.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocat
res, we've 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
--
Th
?
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
--
This email is made from 100% rec
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
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 sa
On Wed, Jul 10, 2013 at 9:16 AM, Matthew J. Roth 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 for the info
ulaw
On Wed, Jul 10, 2013 at 7:40 AM, basteon wrote:
> Hi,
> What codec do you use with yours subscribers?
>
>
> On 9 July 2013 23:45, Chris Gentle wrote:
>>
>> Is there any way I can improve the audio quality in a confbridge in
>> Asterisk 11? I've chang
completely missing something?
--
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
mention that installing Pacemaker/Heartbeat/Corosync or your
other HA solution of preference isn't particularly difficult, and is
agreeably free.
Kind regards,
Chris
--
This email is made from 100% recycled elec
n that deals with blocking?
>
> This is a medical clinic so white-list, black-list is not a good solution
> but it might be good for home use.
>
> Thanks,
> --
> Joseph
>
>
> On 06/13/13 14:30, Chris Gentle wrote:
>>
>> Google the number and you can probably
rd interest charges etc. anybody
> know who it is :-/
>
> --
> Joseph
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar e
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
--
This email is made from 100% recycled electrons
d 11.3.0 on two different boxes with the
same 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
up()
; this is the Local channel that connects to app_ices
exten => 1000_ices,1,Answer()
exten => 1000_ices,n,Ices(/home/asterisk/asterisk-ices-1000.xml)
;exten => 1000_ices,n,Hangup()
;}}}
--
Chris
--
_
--
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
--
This email is made from 100
so 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 configuri
VERBOSE[2544][C-0004] netsock2.c: == Using SIP
RTP CoS mark 5
[May 31 01:47:41] NOTICE[2544][C-0004] chan_sip.c: Call from ''
(188.161.238.232:28203) to extension '011972595595767' rejected
because extension not found in context 'default'.
--
Chris
--
__
annoy your target audience. Recorded calls
always do.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
ble to fix 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
ery impressed. It's certainly my preferred VM platform
at the moment (not just for asterisk stuff, but in general).
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provide
27;re
testing both your upstream's SIP connectivity and also their PSTN
termination.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digit
e was some 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 wrote:
>
mething in asterisk.
Any ideas what I'm missing here? Is there a better way to do this?
--
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live in
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
--
This email is made from 100% recycled elec
en 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
--
This email is made from 100
or evening).
As others have said, the OP might be best advised to request (paid)
assistance with the project on the [asterisk-biz] list.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwi
obile" soft phones we run would not need 911 service
since they also carry cell phones, the soft phones being mainly remote
extensions.
So it sounds like it is at least worth pursuing.
Kind Regards,
Chris
--
_
-- Bandwidth and
or ESINet) does 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:*
e
our emergency traffic. The county seems interested in exploring the
possibility.
So I'm wondering if anyone else has attempted this.
Kind Regards,
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
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
--
This email is made from 100% rec
t 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 u
.
Just my two cents ...
Hope this helps somebody else.
--
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http
Freepbx is over writing the conf files ?
On Mar 15, 2013 8:11 AM, "Luis H. Forchesatto"
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 configured in, but it doesn'
Awesome, thanks. I'll give it a try.
On Mar 11, 2013 4:56 PM, "Joshua Colp" 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
used 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
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Coloc
lear it
up for another day or so. Has anyone else noticed this?
--
Chris
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
SIP endpoint can be had for around 35GBP - I
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,
C
lplan.
Also worth making sure you have retry=0 in your queue config.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Jo
to 7 handsets. In my experience, things start to get a
bit "flaky" above 3 or 4 handsets (specific handsets not ringing
periodically, etc.), so I suspect the base station might be CPU limited
at some point, especially if you're asking it to use an expensive
(computationally) c
s and separate SIP registrations per handset.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a l
sound any good, mind.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory
void having to worry 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
--
This email is made from 100
and see if
that's any better. 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
--
Th
ult for
you to knock up something similar for your property.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Joi
sually 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
--
eractively.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar e
ead 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
--
This email is made from 100% re
pproaches others are using to store voicemail
and recordings, and to make those available across a multi-server
environment.
Let the discussions begin.
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-
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 A
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" wrote:
> This is something I'v
e, then they want it (totally contradicting the meaning of the word
> "business").
>
>
> Christian Savinovich
> VoIP & Telephony Consultant
> 646-982-3572
>
>
>
> Original Message
> Subject: Re: [asterisk-users] DIDForSale spam
> Fro
y the raison d'être for [asterisk-biz]?
Kind regards,
Chris
--
This email is made from 100% recycled electrons
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a l
than 5 seconds.
> I am sure there will be lot of arguments on why you should that and all.
I will refrain myself on any further unproductive communication.
>
> Happy new year to you all.
>
>
> On Wed, Jan 9, 2013 at 4:39 PM, Mitul Limbani wrote:
>
> +1 here.
>
> On Jan
.com
3/8/12
DIDForSale donotre...@didforsale.com via mail.bingotelecom.com
1/9/13
UGH, 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
--
__
On Wed, Jan 2, 2013 at 10:19 AM, Dan Jenkins
wrote:
> On 2 January 2013 16:16, Chris Gentle wrote:
>
>> Does anyone know of any asterisk 11 packages for the Pi? I ended up
>> compiling it myself this weekend. Took a while.
>
>
> Take a look at http://www.raspb
1 - 100 of 3140 matches
Mail list logo
