Anybody have a recommended provider which supports TLS for SIP trunk
communications, or even encryption via IAX?
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I've actually had an AGI script that Asterisk never closed the fork for. It
was testing a particular feature so it was pretty badly written. Ended up
consuming a lot of resources.
No idea why Asterisk hated that script, though. Failed to kill it every
time. But would continue on the dial plan afte
Does anyone have a good contact for their sales? I've attempted calling
their Enterprise sales a few times and was just spun around in circles.
Having a sales rep I can just call would be awesome.
- Logan
On Fri, Jan 4, 2013 at 1:36 PM, Michael L. Young wrote:
> - Original
No problem! Doubt check through a test extension. I don't want to be
entirely wrong. ;)
- Logan
On Dec 30, 2012 12:12 PM, "Geoff Lane" wrote:
> On Sunday, December 30, 2012, Logan Bibby wrote:
>
> > I believe its actually TIMEOUT(absolute)=value. The function name is
Geoff,
I believe its actually TIMEOUT(absolute)=value. The function name is case
sensitive.
- Logan
On Dec 30, 2012 9:53 AM, "Geoff Lane" wrote:
> Hi All,
>
> Asterisk 1.4.22.1 on CentOS 5
>
> I've configured my dialplan to limit the maximum call length on
> ou
I suppose I'm one of the few people that remember the content of threads by
subject and easily catch up...
I'm also on my phone 99% of the time time and the way Gmail lays out emails
makes top-posting beneficial to me.
On Dec 29, 2012 8:57 PM, "Richard Kenner" wrote:
> > I realize the benefits o
m and reads up.
- Logan
On Dec 29, 2012 7:22 PM, "Pete Mundy" wrote:
> On 30/12/2012, Steve Edwards wrote:
>
> > On Sat, 29 Dec 2012, Don Kelly wrote:
> >
> >> 2. How do we change rule #5?
> >
> > -1.
>
> + -1 from me too!
>
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>
> http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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Logan
Logan Bibby, CEO
Ke*o*bi Communications
Tuscaloosa, Alabama
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It is facing the outside world, but I just use SSH's port forwarding. :)
On Dec 4, 2012 10:43 AM, "A J Stiles" wrote:
> On Tuesday 04 December 2012, Logan Bibby wrote:
> > I have a huge logrotate config file and I use Webmin to manage it all.
> >
> > Actually
I have a huge logrotate config file and I use Webmin to manage it all.
Actually, Webmin is a good all-around system management tool, in my
opinion.
On Dec 4, 2012 9:12 AM, "Paul Belanger"
wrote:
> On 12-12-04 10:02 AM, Danny Nicholas wrote:
>
>> IIRC log rotate only rolls the files in /var/log/a
Have you considered using something like Splunk to aggregate your log files
and store a copy for later analysis? Even if you want it to be available to
someone, say a remote customer, via a web panel, I believe you could even
have Splunk put it into another database or make a view in Splunk's
datab
clean
and is a little more flexible and powerful.
- Logan
On Nov 8, 2012 12:41 AM, "martin f krafft" wrote:
> also sprach Paul Belanger [2012.11.07.2340
> +0100]:
> > What is your point of pain? Right now we do most of the
> > configuration, provisioning, and sys
I don't think you can. But you could set it to a lower value like 3 seconds
and give your operators a feature key to pause themselves in the queue if
they need extra work time.
- Logs
On Oct 29, 2012 12:15 PM, "Mitch Claborn" wrote:
> In our sales queue, we have wrapup time set to 15 seconds. W
I had the same problem for a while. I found replacing fax machines with a
scanner and either an email-to-fax program or just web-based faxing had
better results. I don't want to tell you the gateway I used because they
turned out pretty badly in the end. But there is hope!
- Logan
On Oct 4,
l
exten => h,1,Goto(status,hangup,2) ; <- processes a channel not hung up by
the dialplan
On Sep 29, 2012 6:08 AM, "Stefan at WPF"
wrote:
> Thanks Logan. Can you send an extract of your extensions.conf, how you do
> that?
>
> 2012/9/29 Logan Bibby
>
>> I do.
I do. I call the Hangup application in priority 1 so I can send calls there
without needing to call it. Then the h extension goes to status,hangup,2.
- Logan
On Sep 29, 2012 4:36 AM, "Stefan at WPF"
wrote:
> How do you redirect all h calls to your status context? Thanks :-)
>
&
I have a "status" context with a "hangup" extension. All my h calls go
there.
- Logan
On Sep 29, 2012 4:32 AM, "Stefan at WPF"
wrote:
> I have 2 contexts, however both have the same h extension.
> Currently I am doing copy&paste for the h extension -
I agree. A script that read the spool directory, sent enough files to equal
10, wait a few seconds, check again and move more would do the trick.
- Logan
On Sep 27, 2012 11:27 PM, "Patrick Lists"
wrote:
> On 09/28/2012 03:01 AM, Patrick Archibald wrote:
>
>> Hi,
>>
&
Very good point. For revenue critical data like CDRs, being ACID compliant
is important.
MyISAM is compliant. And like InnoDB, can have the features making it
compliant turned off.
On Sep 25, 2012 6:12 PM, "Patrick Lists"
wrote:
> On 09/25/2012 11:18 PM, Logan Bibby wrote:
>
&g
MyISAM would be best, in my opinion. The features that cause the little bit
of performance overhead in InnoDB wouldn't be necessary for CDR storage.
- Logan
On Sep 25, 2012 4:15 PM, "Matt Hamilton" wrote:
> Which one (InnoDB or MyISAM) is preferred for CDR as far as writ
I think a lot of people leave it out in examples for simplicity's sake. It
doesn't instil proper practices in folks' heads.
- Logan
On Sep 24, 2012 12:06 PM, "Eric Wieling" wrote:
> You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on
> the i
Why not use the DIALSTATUS channel variable to determine if a fail over is
necessary?
- Logan
On Sep 24, 2012 6:00 AM, "Thomas Kenyon" wrote:
> I have noticed a peculiar problem recently with the way that the failover
> operates in my dialplan.
>
> I normally have:
he SIP response code.
For my setup, I have an OpenSIPS sever that handles the lower level logic
such as failure routes. I find it a lot amiable to deal with than Asterisk
for that sort of thing.
- Logan
On Sep 23, 2012 5:17 PM, "Jarek Jarzebowski"
wrote:
> Hello,
>
> I need to
.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Logan
Logan Bibby
Ke*o*bi Communications
Mobile: (205) 394-0424
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Hi,
I'm looking to get an ISO of FreePBX or AsteriskNOW installed on a USB that
I can boot from and also be able to save my changes. Is this possible?
My search on web doesn't seem to find anything useful. For now I don't have
the option of having a spare machine or creating a partition on my exi
Hello all,
Thank you for the responses. I really appreciate it.
Since I'm just trying it out for fun, I will begin by using mobile-chan and
see how that goes.
Thanks a lot,
Hitesh
On Wed, Feb 16, 2011 at 4:05 AM, Andrew Latham wrote:
> On Wed, Feb 16, 2011 at 2:49 AM, loga
Hello,
Are there any gateways which allow me to hook a cellphone to Asterisk and
use that line for routing my calls? Basically, I'm looking to play around a
bit and if I can get to connect a cellphone with Asterisk then that would be
great.
Thanks,
Hitesh
PS: I have tried to search on the web, bu
Thanks Paul. Your help is much appreciated here.
> I don't really understand this question - Asterisk can make calls over
> phone lines. And it does it well.
>
Surely, Asterisk does that well, but Asterisk needs to have multiple phone
lines for that. I thought that a traditional switchboard made
Hi Paul,
Thanks a lot for the response.
I'm a novice so pardon me for the stupid questions. I thought that maybe the
PSTN lines don't allow more than 1 simultaneous calls on a line, but on GSM
it might be possible.
I basically want to know how Asterisk can dial out calls from the lines
connec
>
> Hitesh,
>
> The short answer is no. The long answer is that Asterisk (and,
> indeed, any other PBX) is not be able to make or receive any more
> analogue calls than the number of available analogue trunks (or lines)
> to which it has access. You may, however, use an ITSP to send
> outbound ca
Hi,
I'm an absolute newbie and wanted to know the following.
I want to have a setup where I have a PSTN line connected to my
Asterisk box and want to know if it is possible to make more than one
simultaneous outbound call through that VoIP gateway? Can Asterisk do
this magic of concurrent calls o
regulation site dotindia.in <http://dotindia.in> for more
>> information
>>
>> ram
>>
>> On Sat, Sep 6, 2008 at 9:26 PM, logan <[EMAIL PROTECTED]
>> <mailto:[EMAIL PROTECTED]>> wrote:
>>
>> Hello Everyone,
>>
>> Tha
ers for your fxo
> card though, but they usually come with instructions which is quite easy
> to follow.
>
> hth
>
> regards,
> nhadie
>
>
> logan wrote:
>> Hi Jai,
>>
>> If I understand correctly then the DID will enable to call me on the
>> har
hat can come to your asterisk over the internet.
>
>
> Jai
> www.didforsale.com
> *Buy SIP DIDs at low cost unlimited minutes
> http://www.didforsale.com";
>
>
>
> On Fri, Sep 19, 2008 at 9:18 AM, logan <[EMAIL PROTECTED]> wrote:
>>
>> Hello Ram,
&
d, Sep 17, 2008 at 1:10 PM, logan <[EMAIL PROTECTED]> wrote:
>>
>> Thanks a lot Nhadie. I appreciate your help.
>>
>> Could you also suggest some brands or models of the FXO+FXS card that
>> are seamlessly compatible to Asterisk? Also what hardphone I should go
>
Asterisk setup? And which Linux distribution is best suited with
Asterisk?
Thanks a lot for the help :).
Best Regards,
Hitesh
On Mon, Sep 15, 2008 at 12:52 AM, Nhadie <[EMAIL PROTECTED]> wrote:
> Hi Logan,
>
> on your sip.conf you define the sip trunk (which is the voip
Hi everyone!
Okay. I was reading on the voip-info.org about FXO and FXS. Is it
possible just to get a card with FXO and FXS together? I know Digium
sells them, but as I've said, I'm looking to spend too much.
Thanks for everyone'
Philip Edelbrock wrote:
On Nov 15, 2005, at 5:40 PM, Logan wrote:
As stupid as this may seem ::cough::, how do you test to see if
there is voltage on the phone port? Would you plug in a phone that
doesn't require a AC power and runs off the voltage from the phone
line?
Thanks t
was wondering if it was feasable to istall
Asterisk on this box and have that modem (or whatever modem) with a
regular telephone wired to the "Phone" port. I'm hoping to spend very
little (under $50) or none, if possible.
Thanks for
I haven't had similiar experience, but in several threads about sound
quality people have talked about Network cards being the culprit. In
particular, a few people have commented all sorts of problems on
onboard NIC's, since they tend to be of lesser quality than
stand-alone NICS.
On 8/26/05, Adam
If you can get your POTS lines terminated in T1 lines, thats probably
best. Then you can get a digium Quad Card. Otherwise, if you're going
to terminate more than 8 lines, you should probably use a channel
bank.
On 8/25/05, Gulzar Hussain <[EMAIL PROTECTED]> wrote:
> Hi All
>
> I want to terminat
Couldn't you already integrate Asterisk and GoogleTalk with this?
http://www.jivesoftware.org/index.jsp since GoogleTalk just uses XMPP?
On 8/24/05, Andreas Bayer <[EMAIL PROTECTED]> wrote:
> Am Mittwoch, 24. August 2005 10:32 schrieb Mat Stace, Colewood Internet:
> > Lifted from the developer pag
That username & password combination is referenced elsewhere for
different models of ATA's as well. I believe it is somewhat a Vonage
standard.
On 8/23/05, Steve Gladden <[EMAIL PROTECTED]> wrote:
> > There is a fee, but I believe you can call Vonage and get a box
> > unlocked after you are done w
There is a fee, but I believe you can call Vonage and get a box
unlocked after you are done with the service. If I'm remembering
right, the fee is about $10.
BTW, per http://forum.openwrt.org/viewtopic.php?id=1643
try
user: user
pw: tivonpw
For the web interface. You might be able to unlock it f
If you don't mind sharing, what was the vendor that worked great? Thanks!
On 8/17/05, VoIP Newbie <[EMAIL PROTECTED]> wrote:
> I bought 3 from 3 different vendors. One of them has echo issue.
> Another one has an issue regarding PCI master abort. Only one really
> works fine for me. These 3 cards
Yes, but your results may vary. Apparently some people have problems
with "clone" cards (aka regular modems), dropping calls, and having
echos. (Then again some people have reported no problems at all).
E-bay is a good source for these. You can also check out this list
with more information about A
/
Doug Logan
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With 120 Pots lines, why not get 5 T1's, then pick up a couple Digium
cards (A Quad T1, and a single T1 card).
On 8/9/05, Edwin Lam <[EMAIL PROTECTED]> wrote:
> hi folks.
>
> i'm planning to connect * to 120 POTS line. i've done some research
> on FXO cards but unfortunately most manufacturers on
me use, so absolute reliability is not
necissary. As a result I'd like to stay ~20 or less, and get the best
quality I can for this price range. Thanks!
Doug Logan
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You might have better luck posting this question on Asterisk-Dev (on
how to disable checksum etc).
On 8/5/05, Jon Whitear <[EMAIL PROTECTED]> wrote:
>
> >Hi,
> >
> >I'm running Asterisk 1.0.8 on Gentoo, with an X100P (clone) card, and I'm
> >having problems with Caller ID. I have run clidtest, an
Goto hotscripts and do a search for SMS . You will find a number of PHP and
other solutions that will send SMS messages for you. Then you just need
Asterisk to call this PHP, etc. (See Docs on the AGI, or post a comment on the
asterisk-dev list).
>
>Subject: [Asterisk-Users] Send voicemail noti
/opinions, etc? Thanks.
Doug Logan
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It might be helpful if you posted your setup, and relative sections of your
extensions.conf etc.
Is this a new install? are you using VoIP extensions, FX, or what? Is the busy
signal when you call from one extension to the other, when you dial-out? or all
of the above?
>
>Subject: [Asterisk
Page 4 documents the "Authenticate" Feature. I'm a Newbie, so I can't give you
much more help beyond that, but it should point you the right direction.
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-pdf/vm1.pdf
>
>Subject: [Asterisk-Users] How to create a secret
Thanks to everyone who responded. I have a pretty good idea now what we would
need!
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[EMAIL PROTECTED]'s website is http://asteriskathome.sourceforge.net
>
>Subject: [Asterisk-Users] What wrong with asteriskathome.org
> From: "Robert A. Rawlinson" <[EMAIL PROTECTED]>
> Date: Thu, 28 Jul 2005 10:21:18 -0400
> To: asterisk-users@lists.digium.com
>
>I saw on here where there
>
>Subject: [Asterisk-Users] Suggested System Specs - 20 ext, 8 Incoming Lines
> From: Doug Logan <[EMAIL PROTECTED]>
> Date: Thu, 28 Jul 2005 09:59:54 -0400
> To: asterisk-users@lists.digium.com
>
>Hello All,
> We're looking to put in an Asterisk s
If we thought that we
could not make this deadline, and wanted to do outsource this setup to speed
things up, what would be the estimated cost for an installation? Thanks.
Doug Logan
Whitsett, NC
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Hi,
Kib thanks for this - still no luck for me - can you send me more
details of what your setup is?
D.
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t: +44 207 397 8451
m: +44 7966 926694
w: www.bright-talk.com
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Aste
Hi,
Has anyone got this card working with Asterisk? If so what kernel are
you using?
Currently I have installed Fedora Core 3 with the 2.6.10-1.770_FC
kernel
Chan Capi 0.3.5
Asterisk 1.0.7
The diva card is detected by linux and the chan capi is installed in
asterisk
When asterisk
g
it well enough for someone to use in a production environment.
- logan
On Wed, Feb 16, 2005 at 08:06:27PM +0100, Ming-Wei Shih wrote:
> Wang Xiangzhou wrote:
>
> >Sun claims that Linux apps can run on Solaris 10 natively. Is there
> >anyone to run Asterisk on Solaris 10 a
Hi,
Regarding these high bandwidth CODECs - is it possible to upgrade
asterisk to record at a higher quality bit rate too - is Asterisk based
on a 8Khz system. We would like to stream calls from SIP phones to the
internet at a higher quality than a standard phone, also it would be
great to bu
Hi,
Sorry to squeeze 2 issues in here but maybe they are related!!!
I have the following setup:
DECT Phone --> IAXy --> Asterisk Server (with X100p) --> Analogue line
I have 2 issues:
1. Sound quality is not too good from X100p - still get echo - have
tried many gain and echo settings.
Okay, will do. So, is there someone I should send patches to or is
there a process to get cvs write privileges?
- logan
On Thu, Jul 29, 2004 at 02:55:08PM +0900, Sunrise Ltd wrote:
> Logan O'Sullivan Bruns wrote:
>
> > I know Solaris isn't a well tested platform an
Please ignore this message. It turned out my problem was that the
Sipura was falsely detecting the CPC signal. Since doubling the CPC
detection interval time the problem has gone away completely.
Thanks,
logan
On Wed, Jul 28, 2004 at 06:57:01PM -0700, Logan O'Sullivan Bruns wrote:
> I&
ke
some minor code changes to get to compile on my sun box. However, the
fact that almost everything works so perfectly makes me think that it
is a configuration problem not a porting problem.
Again, any advice, things to try and such would be greatly
appreciated.
Thanks,
logan
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