urevoip.com/news.php?rssid=2353
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nna hear it too. Even if this means no solution for me.
> Then I know it's not doable.
:)
Maybe you should read the messages from the list then :)
You've already been replied to.
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ime');
> 'extensions_table` VALUES (3, 'mycontext', '100', n, 'Playback',
> 'my-sound-file');
I'm not sure that's likely to work - or if it does, not in the way you
expect. Likely if you did a query for exten = 100, the n extensions
e
> 139-128=11 there may be a coding mistake). Thus what do you prefer to do? How
> can i examine the core dump file?
http://www.voip-info.org/wiki/view/Asterisk+debugging#CoreSoftwareDebugging
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> configuration with (e.g. /etc/dhcp3/dhclient.conf):
>
> prepend domain-name-servers 127.0.0.1;
>
> Otherwise, your entry in resolv.conf will be overwritten on each DHCP
> lease renewal.
Yeah, although if you're using DHCP, then dnsmasq is possib
asy - bind9 easier I reckon.
To set up on debian do:
apt-get install bind9
add to the top of /etc/resolv.conf
nameserver 127.0.0.1
Then it's done.
Dnsmasq is probably overkill for this type of thing, though some people
in the office prefer it to bind.
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Is there a way to make a virtual extension busy programmatically?
I want to be able to turn lights on and off on a Polycom phone from a script.
-Matt
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itexten(3)
> exten => 3,1,Set(TOTAL=${MATH(${TOTAL}+300,int)})
> exten => 3,n,WaitExten(3)
> exten => 9,1,SayNumber(${TOTAL})
Heh, you might need to say what you're expecting and what you're getting :D
Straight off, all I can see is that 2 does 200, 3 does
, but rather than try and get it
sorted he/she'd rather just bitch.
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http://www
Damn, where were you 6 months ago? ;)
Daniel - Asterisk wrote:
> Just if it is helps someone, based on information at the blog:
> http://allabouthobby.blogspot.com/2009/10/configuring-audiocodes-mp108-mp104-fxo.html
>
> I've summarized the following steps:
>
> *Step 1:*
> Configure audiocodes t
ut nobody seems to get it
> done :-(
Not that I'm aware of - best place to ask would be the IAXClient mailing
list, but I'm pretty sure I'd remember if someone had written one.
Probably the closest would be Tim Panton's work - maybe hunt him down :D
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On Thu, Jan 21, 2010 at 3:30 PM, Jonathan Thurman wrote:
> On Thu, Jan 21, 2010 at 4:56 PM, Matt Darnell wrote:
>> Most manufacturers charge in excess of $80 to upgrade from a 10/100
>> switch to a 10/100/1000 switch built into the phone.
>> The cost might have been in th
don't want to run another cable?
-Matt
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What user are you running Asterisk as?
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On 16/01/10 12:56 AM, nak...@02.246.ne.jp wrote:
> Hi, I have a question about jitterbuffer and PLC.
Do you get the same results if you use:
iax2 test losspct x
Where x is the loss percent you'd like to test?
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Hello,
STDERR goes to the original Asterisk process only, not any "asterisk -r"
connections that you may use. If you launch Asterisk in a "screen" like we
do, then you can see it and log it in context with when the output is
happening. We find it very useful to do it this wa
On 29/12/09 10:22 AM, Leif Neland wrote:
> I want some cheap ip-phones with auto-answer, to work as paging system
> at dinnertime.
> Options, please.
Use some of the Chinese PA1688 or AR1688 phones - support auto answer,
IAX/SIP etc.
Prices around $45
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x27; which should be the incoming or
upload form the clients.
As I am sure you saw, it is not mentioned in the peers and clients section.
Perhaps setting jbforce to no and jbimpl to adaptive.
I am sure you read all that, anyone have any real world experience?
Aloha,
Matt
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> those problems?
Sounds to me like you need to speak with the company providing you a SIP
trunk.
If the calls between VoIP->VoIP are too loud and the calls to PSTN are
too quiet, then likely the provider needs to check their gain settings.
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with Asterisk.
Post your progress as you move through it :)
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http://ww
/asteriskqueues.html and
find out how it's done.
Thanks for reading.
Matt King, CEO Orderly Software.
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you have to do.
--Matt Desbiens
BestVoIPUSA.com
O:603.677.0004
On Sat, Nov 28, 2009 at 4:00 PM, Michael Graves wrote:
> On Fri, 27 Nov 2009 06:50:15 -0800 (PST), bilal ghayyad wrote:
>
> >Hello All;
> >
> >Anyone can advise for the good phone (Polycom, Linksys, ... etc) th
Has anyone seen anything like this? Suggestions?
Is the machine running a GUI? I.E. Gnome/KDE/XFCE etc
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Couldnt you do this by calling MySql? Compare who has the least minutes
used and then send it out the appropriate channel?
--Matt
On Tue, Nov 24, 2009 at 7:07 AM, Eckhard Jokisch
wrote:
> Hi,
> I have 4 ISDN channels (2 lines) and each line may do calls of up to 360
> minutes/month
On 13/11/09 12:33 PM, Tzafrir Cohen wrote:
> On Fri, Nov 13, 2009 at 12:19:54PM +1300, Matt Riddell wrote:
>
>> Maybe the best way would be to make it that the default context only
>> provides the info from the examples unless you provide an option:
>>
>> read_secu
carefully
thought out before proceeding with something which may have a large
impact on new users.
Think what it's like for the 3G video people who have a huge patchset
that they wrote before bringing it up for discussion only to hear it was
the wrong way to do it.
At least the patch is sma
Internet. Call me a cynic. :)
The ignorant won't have changed the default context - they likely won't
even know how to edit a config file - so they're safe.
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gt; what you don't want.
A slight clarification - I wouldn't say it's defences.
By default these calls are sent to the default context (which should not
have the capability to make calls other than test the system).
So, yes you are allowing unauthenticated calls, but to the echo tes
On Wed, Nov 11, 2009 at 1:11 PM, Matt Florell wrote:
> They had a nice booth at Astricon and everything. Haven't heard anything
> about them going down, this might just be an unfortunate IT management
> incident.
>
Both their toll free and fax numbers go to a re-order message
omain servers in listed order:
NS1.NETXUSA.COM
NS2.NETXUSA.COM
They had a nice booth at Astricon and everything. Haven't heard anything
about them going down, this might just be an unfortunate IT management
incident.
MATT---
On 11/11/09, Matt Darnell wrote:
>
> Anyone know what hap
Anyone know what happened to netxusa?
Seemed like they dropped off the web overnight.
-Matt
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you binding to an address that the box doesn't own?
Check the top of sip.conf.
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risk Addons.
> Configuration file: /etc/asterisk/cdr_mysql.conf
Also, the status check is cdr mysql status
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ned to. That way if someone is getting overloaded with
support requests you can move jobs to another staff member.
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iskdocs.org)
2. Set up sip.conf/iax.conf based on what type of softphone
3. Download a softphone - I've listed a few here:
http://www.venturevoip.com/news.php?rssid=2188
4. Make calls :D
The most important step is number 1 - once you get the hang of Asterisk
the rest will be easy :D
-
27.0.0.1/send_sms.php)
Pseudo code for send_sms is:
1. Read AGI variables
2. Get destination variable
3. Include clickatel API file
4. call send_sms function
We also provide an API from our telephone exchanges, but to be fair
you're likely better off just using clickatel yourself :D
On 10/11/09 1:02 PM, Conklin, Tom wrote:
> Have you taken a look at the following?
>
> http://www.astassistant.com/
Also:
http://www.asternic.org
and the newer version:
http://www.fop2.com
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(even from Grandstream).
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http://www.venturev
omer=1234
Or with an Apache rewrite.
It seems that what you're wanting is more on the jabber client side -
you're wanting one that can receive messages and display them as pure HTML.
There may be one - I don't think Adium (the client I use) does it, but
if you
your question.
You're trying to send a link, and what's going wrong?
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http://
used that successfully.
Which brings me to another question - what does Digium recommend people
use on a 1.4 system with their b410p card these days?
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htt
used that successfully.
Ooh really? Where would I find that?
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complicated. Does anyone know if this was
> addressed since 1.2, or can it still happen in 1.4 or 1.6?
Just a shot - all boxes using NTP?
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On 23/10/09 6:11 AM, jonas kellens wrote:
> On Thu, 2009-10-22 at 13:45 +1300, Matt Riddell wrote:
>>
>> It's really simple you just read from standard input and write to
>> standard output.
>>
>> If you tell us a programming language you'd like to use (i
et($s)) {
// Do something with $s
}
(replacing the commented line // with the line in question)
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xtension.
Then if someone dials *79 (or whatever) it picks up the group that the
person dialling *79 is in.
I.E.
* Call goes to Jon (who is in group 3)
* He is away from his desk
* Jane dials *79 (also in group 3) and picks up the call
If Fred (in group 5) were to
?
>>
>> Thanks,
>>
>> Russell.
>
> Require the cell phone user to press a button to accept the call (much
> the same way that the followme app does).
In fact it sounds like what he's actually wanting is the followme app:
http://www.voip-info.org/wiki/view/
/asterisk/dnsmgr.conf
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h based in New Zealand - we're just about ahead of everybody - in
fact it's 1:20 in the morning so I probably should go to sleep :)
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http://
sk/dnsmgr.conf:
[general]
enable=yes ; enable creation of managed DNS lookups
; default is 'no'
;refreshinterval=1200 ; refresh managed DNS lookups every seconds
; default is 300 (5 m
;end"
; and "billsec" may be retrieved inside of of this extension.
endbeforehexten=yes
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ous or threaded
> fashion, unless you specify Asynchronous mode using Async: true”. Guess
> I’ll never be as smart as you, Matt.
:D
I should hope not!!
If everyone was as smart as me, how would I take over the world?
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isk/outgoing, pbx.c would call all of them at once
> without the “first pickup” problem.
Not true - you can use Async mode in an Asterisk Manager originate
command to create a call and return instantly.
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Greetings,
Where can I get the chan_echolink channel driver from? I've seen reference
to it, but have yet to find a place to download/compile it. It is part of
the app_rpt.so module... I am told, but do not see the source with app_rpt.
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cted (after 5-10sec).
1-2 seconds for a query isn't really that quick.
You might also want to do
explain SQLSTATEMENT
where SQLSTATEMENT is the SQL query you are running.
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ad from the standard input (i.e. fgets or similar) and
write to the standard output (printf or similar).
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mple you just read from standard input and write to
standard output.
If you tell us a programming language you'd like to use (i.e.
php/c/perl/bash etc) we can give you a link to some docs and examples.
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On 22/10/09 10:57 AM, das sandesh wrote:
> Hi Matt,
>
> I already used the tuning-primer.sh script to enhance the values for the
> parameters, but still it was being slow to connect when there are lot
> of calls (calls around 150-200 calls). Also I reduced mysql queries in
> t
g name server. In debian just do "apt-get
install bind9" then change your nameserver to 127.0.0.1
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add it? Seems that direct mysql-client code
> should be more efficient than adding ODBC in the middle...
Yep, ODBC would add overhead - you may want to look at using FastAGI and
keeping a MySQL connection open inside your script (i.e. connection
pooling).
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SQL at about 500 queries per second with no problems - we
don't however use Asterisk's MySQL libraries.
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they've done that.
We have a few clients that use Oreka(from OrecX) that does
network-based SIP packet-capture recording. It works very well on
their multi-server setups and the core of Oreka is Open Source.
MATT---
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y run into issues,
so I would recommend moving MySQL/Apache/PHP off to a different server
ASAP.
Thanks for the compliments!
MATT---
On 10/21/09, Robin wrote:
> Hi Matt,
>
> ain't you the vicidial guy? I'm actually trying to get this stuff fixed on a
> vicidial system.
>
?
We usually do not do more than 100 concurrent recordings on a single
server, but we have done up to 250 before successfully.
MATT---
On 10/21/09, Robin wrote:
> Thanks for your response.
> The hardware I have now is not sufficient to set up a ramdisk (just 4 gb)...
> But memory
may even get better sound quality with IAX (less packet loss because
more bandwidth available).
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than you have lines then respond with busy or something.
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ndreds of simultaneous calls at the same time using the Asterisk
Manager.
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Greetings,
I have a fresh asterisk installation. When I install I get all of the
config files. What is the best way to get a 'stripped' down system with
just the bare config files I would need to do a sip connection?
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on manager.conf does not seems to have any
> effect on this issue.
>
>
> Have someone seen something like this before?
It appears you have one action with lots of action ids.
Have you tried running them as separate entries and checking the results
of each action?
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elay?
Next, move to the outbound provider:
1. Are they being sent an invite straight away?
2. Does it contain the full number?
3. What is the length of time between calling and 180 message?
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king the law, go for
it - if you get most of your movies from piratebay then it probably
isn't a problem for you.
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do what you want rather than
get DTMF to trigger it.
I.E. if you want to do a transfer, do a transfer or redirect etc.
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On 5/10/09 3:06 PM, Jerry Geis wrote:
> I found the problem.
>
> The function send_sound() in chan_alsa.c does this:
Best bet is to open an issue on the bugtracker and post your patch
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t with 5 calls per second.
Also, I don't notice anything to set it to this, are you sure you're not
trying to start all those calls concurrently?
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the console...
How are you spacing the calls out? I.E. how many calls per second?
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On 2/10/09 12:41 AM, das sandesh wrote:
> Hi Matt,
>
> When I get can more that 150 calls, i get a busy signal (Congestion) for
> the calls above 150 - says "your call cannot be completed now", its
> allowing only 150 callsIs there any thing related to field
> des
options?
Are you able to get a packet dump of how the packet should look?
If so, you may be able to add a sip header if necessary.
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more than that active on a machine.
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the conference room, but person A is
> disconnected.
Is there an extension:
dynamic-nway,282,1
Oh, and please refrain from using HTML emails to lists.
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ected.
Why doesn't A just call the number they've been transferring people to?
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gt;
> Country numbering plan can be easily found.
>
> Anything finer then that and you will need to pay.
That link I provided is correct at least for New Zealand cities etc
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"]?200)
> exten => _X., 200, Hangup
>
1. works
2. should work
3. works
4. r option is horrible, but works
5. works
6. weird but works - it's going to go to 7 anyway, so effectively this
line does nothing
7. again, r is horrible but works
8
n => _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
1, 2, yep.
> [to-test]
> exten => _X., 1, SetCallerPres(allowed)
> exten => _X., 20, Monitor(wav,/tmp/test-${UNIQUEID},mb)
Normally 2 comes after 1 rather than 20 - looks like you're missing
/www.itu.int/oth/T0202.aspx?parent=T0202
Bear in mind that these numbers change reasonably regularly.
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n dialed,
and letting you receive calls from both the PSTN and your VoIP provider
without the need use two separate telephones. "
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m phones.
It's not really that difficult as long as you understand the tftp
provisioning for the phones.
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On 17/09/09 1:57 PM, Jeff LaCoursiere wrote:
>
> On Wed, 16 Sep 2009, Doug Lytle wrote:
>
>> Matt Riddell wrote:
>>> Basically, the phones are displaying 79 on the screen (the number the
>>> dial for pickup) - as you'd expect, but they'd like to
play something back.
Do you get an error in the console?
What do you have in /etc/asterisk/modules.conf
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(the number the
dial for pickup) - as you'd expect, but they'd like to see the CID of
the person who called in.
I've managed to get the CID passed along during subsequent transfer of
the call, but would love to be able to provide them with the icing on
the cake :)
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fully
compatible with QueueMetrics.
MATT---
On 9/16/09, Maria Cristina Bayno wrote:
>
> Hello Team,
>
> IVR selection of QUEUEMETRICS
>
> As we know queuemetrics had an IVR selection functionality where it can get
> the IVR keypress of a caller.
>
> We saw this link
t; experience you get is my right
If you're worried about performance (I assume CPU usage as both will
sound the same) why don't you get a transcoder card - I think both
companies do one.
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I
didn't understand it either, but there it is.
MATT---
On 9/14/09, John A. Sullivan III wrote:
> Hello, all. I see there is an "o" option for the Dial() command which
> reverts to the previous behavior of using the original callerid
> throughout the call - I suppose m
not connected to Asterisk and Asterisk just sees all
calls as origination from your proxy, surely the place to sort this out
would be the proxy.
Can you not set a variable in the proxy before sending the call to
Asterisk and use the sip header function to retrieve it once in As
rom the asterisk-generated cdr-s - I don't really want to start
> relying on them)
For every billable item we use a code for the account and store it in...
accountcode :)
--
Cheers,
Matt Riddell
Director
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talled the ia32-libs package as well, but no success.
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Cheers,
Matt Riddell
Director
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http://www.venturevoip.com/c3.php (C
not sure I've seen many bounty requests
in the last few years - probably because most of them never made it to
fruition.
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Cheers,
Matt Riddell
Director
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http://www.venturevoip.com/s
if someone didn't ask the manager for
more time.
We actually used the SendText function to print a message on the screen
of the phone showing the remaining credit (and wrote a patch for
Asterisk so you could do this from the manager - included in 1.2 if I
remember correctly).
--
Chee
rSend(asterisk,m...@jabber.venturevoip.com,Support Call
From ${CALLERID(num)})
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Cheers,
Matt Riddell
Director
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On 10/09/09 10:53 AM, David @ULC wrote:
> Asterisk-Addons : No
Well, that's a start - you'll need to at least install the MySQL CDR addon.
:)
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Cheers,
Matt Riddell
Director
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