mit option in 1.6/1.8?
>
> The correct answer is to use ringinuse=no in queues.conf and callcounter=yes
> in
> sip.conf.
>
Leif,
Isn't callcounter for 1.6 and not for 1.4?
-Matt
--
_
-- Bandwidth and Colocatio
ty good wake up call to us to really start
locking things down.
I know its lame, but from Network Security Hacks.
Security isn't a noun, it's a verb; not a product, but a process
--Matt
On Fri, Oct 15, 2010 at 11:50 AM, Jeff LaCoursiere wrote:
> On Fri, 2010-10-15 at 11:20 -0400, St
nd setup hints in your extensions.conf for each
> peer.
Warren,
Setting the call limits was my issue. I am on a test machine and
didn't have it set. Thanks for the help!
-Matt
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00b s...@macro-tl-userexten Up VoiceMailMain(101)
1 active channel
1 active call
'core show channels' show SIP/101 is use but 'queue show' does not.
-Matt
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http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AddQueueMember
Thanks for the help.
-Matt
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wrote:
> What version of asterisk are you using and method are you using to login your
> agents? I recently had this issue with a 1.4
ive queue calls.
Is there a way to stop this from happening?
-Matt
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htt
rs now and
have had nothing but the best with them. The only quirk that i'm still
looking into, is that dang Intercom button. Other than that, Grandstreams
are really the way to go IMHO.
Side note: We've probably got close to 400 deployed
--Matt
On Wed, Oct 13, 2010 at 10:43 AM, Br
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas wrote:
> One more thing: Make sure that the port going to your data-DHCP server
> doesn't have the voice VLAN set on it. I troubleshot an installation for a
> few hours before thinking of this...
Interesting, the DHCP server for the voice and d
en, so yep (at least as of last
time) it is a dead project.
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Matt Riddell
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http://www.vent
ing a "logger reload" in
the console?
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Matt Riddell
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http://www.venturevoip
On Wed, Sep 22, 2010 at 10:46 AM, Adam Moffett wrote:
> That's probably what I'm going to have to do. Thanks.
>
> > I suppose that merely removing ATA and asterisk from the middle, and
> > plugging a pots line into a fax machine is out of the question.
> >
> >
>
>
>
If you are running 1.6, you c
isk.
If you have any ideas or suggestions, feel free to mail me on them.
Oh, and we've moved the Daily Asterisk News web server to Dallas, TX, so
it should be a bit quicker for those of you in the states - well,
anywhere except New Zealand really :)
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Matt Riddell
__
ge cut, you'd see much better success
from it, because people would redirect there.
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Matt Riddell
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ng various people's
problems via firmware upgrades pretty regularly. I'd advise flicking
the people over at Snom and email, then posting back here with your
results :)
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Matt Riddell
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imes it works fine and i do not encounter this problem. But it
> happens
> very regularly, too often i would say.
Try it with Zoiper and see how you go. I've not seen the same thing happen.
It may also be that you are using qualify and that the peer is too far away.
What do you get when
using for the Queue?
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Matt Riddell
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oritization and a web-based agent screen that easily
integrates with web-based CRM systems. It is also Open Source and has no
licensing costs:
http://www.vicidial.org
MATT---
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On 3/09/10 8:32 AM, Arnaldo Giacomitti Junior wrote:
> There´s a way to get the channel signalling in dialplan?
>
> I have changed the code in channels/chan_dahdi.c and includes:
Upload it as a patch to the issue tracker:
http://issues.asterisk.org
--
Cheers,
Mat
Just a heads up. It would appear that Vitelity is back online and
processing calls and the portal is back up and running.
On Sat, Sep 4, 2010 at 12:14 PM, Matt Desbiens wrote:
> Not that I'm aware of short of our direct contact. It would appear from
> the traceroutes that i&#
To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--Matt
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] NOTICE[27507]: chan_sip.c:15679 sip_poke_noanswer: Peer
'vitel-outbound' is now UNREACHABLE! Last qualify: 1193
[Sep 4 10:26:23] NOTICE[27507]: chan_sip.c:12528 handle_response_peerpoke:
Peer 'vitel-outbound' is now Reachable. (176m
27;t need to have the @192.168.0.1 in there - just make sure the
username and password are correct in the user's device.
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Matt Riddell
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http://www.venturevoip.
ions. The results were that incoming calls
> would not hear any ringing tones and the call would be ended by Teliax
> after 21 seconds.
You could just answer the call before dialling your internal extensions.
--
Cheers,
Matt Riddell
___
htt
few seconds of the call, I lose audio from the
> asterisk box to my soft phone, but not the other way around. This looks
> like one commit, but obviously I would like to know what's going on
> here?
What's in the commit?
--
Cheers,
Matt Riddell
__
On 25/08/10 7:35 PM, Julian Lyndon-Smith wrote:
> Hey Matt, thanks for the response.
>
> I know it sounds impossible. Hell, I sound like a user :) But it *is*
> happening. And only on the cisco phones. We're trying to lab it up
> right now. What should I be looking for in t
Verbose(-- CID is<${CALLERID(num)}>)
> exten => jw,n,PrivacyManager(3,10)
> exten => jw,n,GotoIf($[${PRIVACYMGRSTATUS}=FAILED]?bad)
> exten => jw,n,Verbose(-- CID is<${CALLERID(num)}>)
> exten => jw,n,Dial(SIP/1000,60,w)
Maybe you could do:
Set(CDR(userfiel
VoIP traffic in New England into a tailspin.
--Matt
On Thu, Aug 26, 2010 at 3:23 PM, Andres wrote:
> On 8/26/2010 2:55 PM, M S wrote:
> > Hi,
> >
> > I've been getting complaints lately that callers to my IVR are
> > pressing a digit once but the system is respo
Has anyone successfully compiled the AMR codec into an Asterisk install, and
if so, what steps did you take?
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priority scheduling is the culprit. If it's something else, like
> memory swapping, there's nothing the canary can do to fix that.
Aha, explains why I've never seen the canary die :D
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Matt Riddell
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On 25/08/10 7:14 PM, Tino wrote:
> Yes, we need to record the message
:D So use the Record() application :D
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Matt Riddell
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and see what turns up. Also make sure it's 100%
repeatable :D
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Matt Riddell
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gt; SELECT COUNT(*) FROM cdr WHERE billsec>duration;
+--+
| COUNT(*) |
+--+
|0 |
+--+
1 row in set (0.31 sec)
mysql> SELECT COUNT(*) FROM cdr;
+--+
| COUNT(*) |
+--+
| 190052 |
+--+
1 row in set (0.
On 20/08/10 1:52 AM, Tino wrote:
> Hello,
>
> Is there a way to capture the answering machine message when the dialer
> detects the answering machine.
Record?
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Matt Riddell
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ost.
Do you have high priority enabled in asterisk.conf?
Are you using DAHDI? (maybe kernel not happy)
I really thought that the canary should have sounded if Asterisk got in
a loop - or maybe that only happens with high priority?
--
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Matt Riddell
_
ve transfers, transfer
; only media stream
Well, it depends on what version. The above is from a 1.4 system -
earlier systems had notransfer=yes, but not the mediaonly option.
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Matt Riddell
___
If anyone has any info on this it'd be much appreciated - haven't found much
about this topic anywhere. We are setting up sip probe monitor to make sure
that our Asterisk boxes are up and functional (or at least responding to the
sip protocol) and we need to determine the appropriate probe syntax f
Has anyone successfully implemented Asterisk as a voicemail server for a
GSM/cellular system and worked out a way to send notifications of new
messages to the phones?
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5551234 (or whatever their DDI number is) and
priorities which increase. It makes some things a bit harder, but you
can always use labels and the read application.
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Matt Riddell
___
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to consider that as ALAW.
That would be signed linear (SLIN).
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Matt Riddell
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http://www.venturevoip.com/st.php (
info?
:) Take it you don't use Google Analytics, Facebook insights,
Feedburner, Amazon EC3 etc etc.
Sure you have to decide who you want to trust (personally I trust the
humbuglabs guys) and what their level of protection is (are they looking
after their own security), but it seems to be th
>
>
>
>
> >I've seen caller-id come through from carriers as:
> >NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
>
> >My question is: what is the correct way to send Caller-ID by set
> standards?
>
>
>
> The correct answer to this depends on where you are.
>
>
> IMO the answer would be #2, but #3
Continental US-48.
On Mon, Aug 9, 2010 at 3:36 PM, Danny Nicholas wrote:
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt
> *Subject:* [asterisk-users] Correct Caller-ID
>
>
>
> >I've see
I've seen caller-id come through from carriers as:
NPA-NXX-, 1-NPA-NXX-, and +1-NPA-NXX-
My question is: what is the correct way to send Caller-ID by set standards?
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eny, I did it in conjunction with insecure set to
>>> port,invite to allow gateways that didn't register and don't use
>>> username/secret to originate calls but only from the ip range in
>>> permit. In fact it was for a provider that had gateways on a large
&
Steve,
Can you recommend any 3G femtocell to VoIP manufacturers? I'm coming up
very dry. OpenBTS sounds like it would work, but is way too expensive to
roll out to residential homes.
On Mon, Aug 2, 2010 at 6:53 PM, Steve Kennedy wrote:
> On Mon, Aug 02, 2010 at 03:36:59PM -0400, Ma
#x27;m not the first one to try and write this type of IVR, so
> would appreciate any feedback on writing this.
http://www.voip-info.org/tiki-index.php?page=Asterisk+tips+Wake-Up+Call+PHP
--
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Matt Riddell
___
http://www.venturevoip.com/
and why?
Late response, and I don't use Windows any more, but SecureCRT with
tabbed SSH windows and buttons which can be set up for things like "nano
/etc/asterisk/extensions.conf" make life pretty simple.
On Mac I now use iTerm (similar thing).
--
Cheers,
Mat
On Mon, Aug 2, 2010 at 3:53 PM, Steve Totaro wrote:
>
>
> On Mon, Aug 2, 2010 at 3:36 PM, Matt wrote:
>
>> Is anyone aware of a GSM femtocell that will trunk back to a VoIP
>> softswitch such as Asterisk?
>>
>>
> I have not, but I have had great luck wi
Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch
such as Asterisk?
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,
MATT---
On Sun, Jul 25, 2010 at 8:29 PM, Juan David Diaz wrote:
> The only big difference I know, is:
>
> VicidialNow - *based on CentOS* - Vicidial 2.0.5.1rc1
> ViciBox - *Based on OpenSuse* - Vicidial 2.0.5
>
> The core of the call center for both of them is Vicidial.
>
&g
It's not necessarily this simple. There is an approximately 50-75foot cable
run through ceilings and walls (CAT5) to the location where the phones will
be. At the phone location there is no power.
On Thu, Jul 22, 2010 at 11:33 PM, David Backeberg wrote:
> On Thu, Jul 22, 2010 at 2:46
st phones are class
> 3 devices. The math just doesn't work out. Even if you used the draft
> standard for class 4 (~30W), you could still power max 2 devices at 15W/ea.
>
> -Dave
>
> On Thu, Jul 22, 2010 at 2:46 PM, Matt wrote:
>
>> I've got an interesting situa
I've got an interesting situation where I have one cable run from the feed
area to the service area. I have three devices that I need to power at the
service area. Is anyone aware of a device that will take the POE from the
cable run and then allow me to split it to two or three devices at the
s
Has anyone used FreeSide to do billing with Asterisk?
How easily did you find it integrated?
With external systems?
With your credit card processor?
How easy was it to add additional fields or 'service' types?
--
_
VoIPInnovations from what I understand is pretty good, haven't dealt much
with them though. Worth a call and an interop.
--Matt Desbiens
//EOF
On Thu, Jul 8, 2010 at 3:33 PM, Adam Moffett wrote:
> I'm in the Northeast US and looking for any recommendations on Level3
> resell
window by accident more times than I can count,
most of the time its from doing a right-click expecting a context menu to
get a 'copy' action and then i just end up pasting what i actually wanted to
copy :|
--
Matt
--
Matt
--
__
I've noticed from time to time, that fail2ban just craps out, so, this might
be of interest to the community assuming you use 192.168.100.0/24 on your
network
iptables -A INPUT -s 192.168.100.0/24 -j ACCEPT
iptables -A INPUT -s carrierip.x.x.x -j ACCEPT
iptables -A INPUT -s 127.0.0.1 -j ACCEPT
On Wed, Jun 30, 2010 at 4:26 PM, Ryan Wagoner wrote:
> On Wed, Jun 30, 2010 at 6:10 PM, CunningPike wrote:
>> On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell wrote:
>>> Thank you Andrew,
>>>
>>> I will check it out. We are currently running 1.4.
>>>
&
On Wed, Jun 30, 2010 at 12:10 PM, CunningPike wrote:
> On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell wrote:
>> Thank you Andrew,
>>
>> I will check it out. We are currently running 1.4.
>>
>> -Matt
>>
>> On Mon, Jun 28, 2010 at 3:48 PM, Andrew La
client
to pull the recording and all associated call information... I know its a
long shot and everything should be in SQL to be pulled from the DB and
posted, but I want to know what I'm getting into before I dive in...
-- Matt
//
Thank you Andrew,
I will check it out. We are currently running 1.4.
-Matt
On Mon, Jun 28, 2010 at 3:48 PM, Andrew Latham wrote:
> Remote Party ID in trunk, it works There are hacks for other versions.
>
>
> ~
> Andrew "lathama" Latham
> lath...@gmail.com
>
distribution of Asterisk.
To download your free evaluation version, and start benefiting from
improved visibility and easy call center management today, just visit
http://www.orderlyq.com/asteriskcallcenterstatistics.html
Thanks for reading.
Kind regards,
Matt King
Managing Director
Orderly
have ever tested do this, Polycom, Linksys,
Cisco, Grandstream, Yealink, etc.
-Matt
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ROUP() and GROUP_COUNT() functions in the dialplan to enforce call
> limits.
But why 'callcounter', it is frustratingly close 'call-limit' and
there is no possible way to use logic to determine what it does.
If a change was to be made,
aying "Please
hold while you are transferred to my cell number".
--
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Matt Riddell
Managing Director
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omplete type
> chan_ooh323.c:1938: error: dereferencing pointer to incomplete type
> chan_ooh323.c:1940: error: dereferencing pointer to incomplete type
> chan_ooh323.c:1943: error: dereferencing pointer to incomplete type
Do you need OpenH.323?
If not, run
make menuconfig
and disable
designed for important data (i.e. where accuracy is more
important than timing)
--
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Matt Riddell
Managing Director
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nging up, but I don't have a safe way to press buttons.
Why not just use followme for everything but the car, and if that fails,
send the call to the car normally?
--
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Matt Riddell
Managing Director
___
http://www.venturevoip.com/ne
you create an account on issues.asterisk.org you can file a
disclaimer for the code and then the patch can be added to the base
Asterisk install (assuming it meets coding guidelines etc).
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Matt Riddell
Managing Director
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All - I have an AA50 appliance that has been running on my 25 user
network for the past year. I need to change the IP address of this
device for administrative purposes. Are there any caveats that I need
to be aware of or do I simply need to change the address and reboot my
phones? Thanks-
--
RQ and set processor affinity for all the IRQs to
particular cores... so far I haven't had kernel pancs since doing this, but
its still a little too early to say if it has fixed the issue 100% or not.
--
Matt
On Mon, Mar 29, 2010 at 8:30 PM, James Lamanna wrote:
> Hi,
> I'm trying to figu
on your server or a router between the
phones and Asterisk
server?
--
Matt
On Wed, Mar 24, 2010 at 7:07 PM, Danny Dias wrote:
> Hello my friends...
>
> Currently we are using the following firmware versions on ours aastra 55i:
>
> Firmware Information
> Attribute Value
&g
Just try running:
asterisk -vcd
And you'll see the error.
Alternatively you can edit /etc/asterisk/logger.conf to allow you to
have a full log.
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Matt Riddell
Managing Director
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OLONG-5500
Alternatively your threshold might be too high - do a few tests to your
own phone and make sure it recognizes the individual words.
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Managing Director
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:TOOLONG-5500
Looks like it's missing the first word - some VoIP providers take a
while to pass audio - might be that there is a delay in your dialplan or
that the first words of audio are simply not transmitted.
--
Cheers,
Matt Riddell
Managing Director
d
accepts aLaw, then that's likely what you want to use (bear in mind that
they might still use an upstream provider who uses G.729 etc).
Easiest option is to just choose aLaw or uLaw based on your country.
--
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Managing Director
___
en clicked the
download link.
When done from an iPhone it brings up the app with a link to download,
on my Mac it opens iTunes to the application page.
--
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Matt Riddell
Managing Director
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On 20/03/10 9:47 AM, Tzafrir Cohen wrote:
> On Fri, Mar 19, 2010 at 10:50:17AM -0400, Zeeshan Zakaria wrote:
>> Hi Matt,
>> This is very useful. But what about android platforms? Will it run on it?
>
> Just use an RSS reader. I guess browsers and RSS readers on the iPhone
>
On 19/03/10 1:19 PM, Adrian Marsh wrote:
> Hello,
>
> I’m looking for some advice on securing Asterisk.
Have a look at fail2ban:
http://www.voip-info.org/wiki/view/Fail2Ban+%28with+iptables%29+And+Asterisk
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Managing
rovide an option (r IIRC) to provide ringing
instead of music on hold.
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Managing Director
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Hi all,
I've released another free app for the iPhone and iPod touch - this one
lets you read the Daily Asterisk News.
Hope you enjoy it :D
http://www.venturevoip.com/news.php?rssid=2371
--
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Matt Riddell
Managing Director
___
Just as an FYI, your 1.4.29.1 patch applies successfully against 1.4.30 as
well. I've got a patched 1.4.30 system compiled and ready to install later
tonight during off-hours and will begin having people test tomorrow. People
here are going to be quite thrilled about having T.
l-setup stuff through an AGI file -
'dialparties.agi' and I'm not sure how good/bad its logging is.
Have you considered simply upgrading to the latest fpbx release? 2.7 was
released about a week or 2 ago.
--
Matt
--
__
Hi Dave,
Thought I'd give you an update - I completely rebuilt my astdb the other
night by renaming it, having * recreate it and then re-creating all my
custom entries in it.
Didn't have any effect, I had somebody report false MWI notifications again
earlier this morning.
--
Matt
On
really be a problem...
Again, looks like you have the order of the channels round the wrong way.
If you originated to a SIP device and sent the other end to the
application PlayDTMF, then it would be sent to the SIP device (if that's
what you want).
--
Cheers,
Matt Riddel
it first though.
I would however like to believe that if * is no longer supposed to be using
berkdb for any VM reference data, that any calls to read the voicemail
counts from the DB should have been removed.
--
Matt
On Mon, Mar 8, 2010 at 5:08 PM, Dave Poirier wrote:
>
> So a couple of questi
I just downloaded a copy of this, by any chances does Zoiper by any chance
have diff files available for a more recent 1.4.x release? (I know 1.6 is
probably out of the question)
Thanks,
--
Matt
On Mon, Mar 8, 2010 at 12:11 PM, Matt Watson wrote:
> Awesome!
>
> I was an Attrafax cus
at t38
gatewaying.
--
Matt
On Sun, Mar 7, 2010 at 4:52 AM, Zoa wrote:
>
> On friday we finally released Attrafax under a GPL2 license.
> It comes with its own set of modems and built in transparent gatewaying.
> The solution should be quite stable as long as the line quality is ok.
&
Tilghman,
That's what I finally figured out... my understanding, though, is that it is
preferred to use 'file' over mpg123?
On Sat, Mar 6, 2010 at 1:32 AM, Tilghman Lesher wrote:
> On Friday 05 March 2010 17:19:06 Matt wrote:
> > For some reason I have to set the type t
For some reason I have to set the type to 'files' if I set it to 'quietmp3'
I get nothing, even though the files are valid MP3 files that play on
another asterisk system... does that mean I've got something installed
wrong?
2010/3/5 Håkon Nessjøen
> On Fri, Mar 5, 2
Forgot to include: I'm running 1.6.2.5
On Fri, Mar 5, 2010 at 5:29 PM, Matt wrote:
> I'm trying to setup my asterisk system for the least overhead as possible.
>
> My understanding (and experience with other systems) leads me to believe I
> can run any MOH using a certain
I'm trying to setup my asterisk system for the least overhead as possible.
My understanding (and experience with other systems) leads me to believe I
can run any MOH using a certain class through a single 'player' as opposed
to starting an independent stream for each MOH instance. However, try as
I've got a ton of files in doc but not that file.
On Thu, Mar 4, 2010 at 7:32 PM, Leif Madsen wrote:
> Matt wrote:
> > Already found it -- but I was under the impression this was deprecated
> > and removed in 1.6?
>
> Try looking in the doc/ subdirectory of your Ast
lad I;m not alone on this one!
I;m more than happy to do any testing of patches if anybody has any
suggestions.
--
Matt
On Tue, Mar 2, 2010 at 1:36 PM, Dave Poirier wrote:
> We are having an issue with Asterisk 1.6.1 and the MWI turning on when a
> user doesn't have voicemail.
Steve,
Already found it -- but I was under the impression this was deprecated and
removed in 1.6?
On Thu, Mar 4, 2010 at 6:44 PM, Steve Edwards wrote:
> On Thu, 4 Mar 2010, Matt wrote:
>
> > I'm trying to setup a situation where I have agents on POTS lines at
> remote
>
I'm trying to setup a situation where I have agents on POTS lines at remote
locations. I want to allow them to call a DID, log into the Asterisk
system, and be an agent. Ultimately I'd like Asterisk to call them at the
number they were at when they logged in.
Does this functionality exist in As
mmy Botten Jensen wrote:
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> Matt Riddell skrev:
>> Yeah, the problem's not the origination.
>>
>> The problem is that calls originated asyn with accountcodes show up
>> in
>> show channels concise w
ate
than Asterisk can fill out the details.
Apologies for top post, laptop is running a defrag.
On 24/02/2010, at 9:32 AM, Tommy Botten Jensen
wrote:
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> Olle E. Johansson skrev:
>> 23 feb 2010 kl. 20.18 skrev Matt Riddel
The responses from the Asterisk manager on your machine start
providing responses of no account code when calls are initiated at a
higher rate.
On 24/02/2010, at 12:59 AM, CDR wrote:
> My dear friend Matt Riddell insists that the Manager only can dial 5
> calls per seconds, which
Also, why are you saying your name is Philip?
On 24/02/2010, at 12:59 AM, CDR wrote:
> My dear friend Matt Riddell insists that the Manager only can dial 5
> calls per seconds, which I find ridiculous. Is there a way to prove
> him wrong and have him lift the limit that has been
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