Same activity from these IPs:
174.129.137.135
89.35.123.12
209.20.66.234
184.73.30.42
184.73.44.61
87.106.187.137
194.44.244.187
203.55.198.100
209.76.47.11
94.74.229.229
93.184.79.59
209.62.53.242
On Thu, Jul 1, 2010 at 10:56 PM, Jamie A. Stapleton <
jstaple...@computer-business.com> wrote:
>
On Thu, 1 Jul 2010, Kyle Kienapfel wrote:
> For embedded systems, I plan to try running asterisk off of an old AMD
> Geode based thin client, It's got debian but I've replaced /sbin/init
> with a link to busybox and I have a script that does the bare minimum
> to get operational and have syslog lo
On Thu, Jul 01, 2010 at 10:19:08PM -0500, Karl Fife wrote:
> Calls that come in on DAHDI FXO ports are routed to [context], extension 's'
>
> INSTEAD, I would like to route specific ports to specific extensions, For
> example:
>
> I want DAHDI/1-1 to go to 1234
> I want DAHDI/1-2 to go to 2345
>
Calls that come in on DAHDI FXO ports are routed to [context], extension 's'
INSTEAD, I would like to route specific ports to specific extensions, For
example:
I want DAHDI/1-1 to go to 1234
I want DAHDI/1-2 to go to 2345
I want DAHDI/1-3 to go to 3456 ...etc
What is the CLEANEST way to do this
On Fri, Jul 2, 2010 at 1:07 AM, Kyle Kienapfel wrote:
> http://www.skype.com/intl/en-us/business/skype-manager/
> "Currently, we're expecting a suggested charge of between €2 to €10
> per seat/month."
>
> Whoops, *grabs a napkin*
And that is in addition to the per channel charge of SfS !
/r
--
hi, all
recently, i face a GotoIfTime problem
GotoIfTime("08:00:00-07:00:00,mon-sun,*,*?95040263008,start")
as you can see the section is 08:00:00-07:00:00 , which is the begin
time is later than the end time
what's this refers then?
in my test , my system time is 10:57:00, but this check
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina wrote:
> I've experienced a similar DTMF issue with recent asterisk 1.4 versions
> (1.4.32, 1.4.33.1), I'm not sure about 1.6.2.X. What happens here is
> that the DMTF activated features, like disconnect (default *) or blind
> transfer (default #) stop
John Ervin wrote:
> Where are the rules for posting in this discussion group? Just curious.
>> It's a rule on this list, although it's frequently ignored.
>
>
Further searching shows there is NO written rule regarding top bottom or
even sideways posting
http://www.asterisk.org/community/rules
Wh
John Ervin wrote:
> Where are the rules for posting in this discussion group? Just curious.
>> It's a rule on this list, although it's frequently ignored.
>
>
One might go here :
http://www.asterisk.org/support/mailing-lists
But the link to "rules" is broken!!
John Novack
--
Dog is my Co-
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=230492577678#ht_500wt_1076
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
Where are the rules for posting in this discussion group? Just curious.
It's a rule on this list, although it's frequently ignored.
smime.p7s
Description: S/MIME Cryptographic Signature
--
_
-- Bandwidth and Colocation
El 29/06/10 15:28, Mark Deneen escribió:
> We are experiencing intermittent DTMF problems here, with the
> following setup:
>
> ITSP -> PIX -> Asterisk (g729, RFC2833 for DTMF).
>
> I am running Ubuntu server 10.04, but Asterisk is compiled by us and
> not installed from the software repository.
On Thu, Jul 1, 2010 at 3:50 PM, Kyle Kienapfel wrote:
> On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere wrote:
>>
>> On Wed, 16 Jun 2010, Randy R wrote:
>>
>>> On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere
>>> wrote:
>
> pretty much giving up on Skype for Asterisk (and Skype for SI
On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere wrote:
>
> On Wed, 16 Jun 2010, Randy R wrote:
>
>> On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere
>> wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be charging a monthly fe
For those codecs an interfaced DSP might be the only option due to
lack of, or expensive software options.
I had an easier time looking into MELPe than I did with CVSD, so I
looked around just a little bit to satiate my curiosity
https://docs.google.com/viewer?url=http://www.compandent.com/MELPeP
The IP 69.175.35.186 has just been banned by Fail2Ban after 293 attempts
against our server.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Timms
Sent: Thursday, July 01, 2010 11:32 AM
To: Asterisk Users Mailing List - Non-Commer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Thursday, July 01, 2010 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote Party ID is
Sorry to answer my own question here - had a look at the headers of
Tilghman's last email and it contained this:
" X-message-flag: Major security vulnerability detected! You should shutdown
your computer immediately and upgrade to Ubuntu Linux 8.04 or
later."
Cute. Leaving aside the fact
As an interesting aside, every email I get on this list coming from Tilghman
Lesher is marked with a "To Do" flag by my email client. Every single one.
I don't have any inbound filter that would explain the behavior either.
On 7/1/10 1:15 PM, "Tilghman Lesher" wrote:
> On Thursday 01 July 2010
On Thursday 01 July 2010 15:01:45 Danny Nicholas wrote:
> On Thursday, July 01, 2010 2:37 PM, Adam Moffett wrote:
> > Steve Howes wrote:
> > > DON'T reply to people off list. And stop bloody top posting.
> >
> > Is bottom posting your personal preference or is that a rule on this
> > list? I have
TY, John.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Novack
Sent: Thursday, July 01, 2010 3:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Remote Party
> When I dial a peer with no registration, SIP channel currently waits
> many seconds before returning ${DIALSTATUS} "CONGESTION" - how can I
> shorten this timeout?
Look at qualify=yes for that peer.
Use ChanIsAvail() before you dial.
Use SIPPEER(peername|status) to check registration status.
Use
A religious argument that will not be resolved or go away
Top posting to some doesn't work because of their mail clients
Bottom posting is a PITA to many because some don't trim off signatures
and other un-necessary text.
Much archive space and bandwidth is wasted on this subject, which will
not
BP is not a RULE and I wish people would STOP BITCHING about it. If you use
MS Outlook to reply to this list, TOP POSTING is the default behavior. If
someone wants to write a nice "how-to" on Bottom posting in MS Outlook, I'll
be happy to read it.
-Original Message-
From: asterisk-users-
> DON'T reply to people off list. And stop bloody top posting.
>
> Steve
>
>
Is bottom posting your personal preference or is that a rule on this
list? I have personally always found top posting easier to follow
because the newer content is at the top.
--
_
-Original Message-
From: Ryan Wagoner
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thu, Jul 1, 2010 6:19 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On Thu, Jul 1, 2010 at 11:52 AM, wrote:
Thanks a lot.
Applying
Hi, we've just been able to find the problem. Apparently it was related to the
softphone. We've installed another one and the call is performed ok.
Thanks!
Anahi Ludueña
From: a_ludu...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Wed, 30 Jun 2010 19:59:14 +
Subject: Re: [
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hey Guys,
Saw your article at www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx
and had a question regarding the directory on a 7960 POS3-08-6
not running call manager.
I quickly figured out each directory only holds 32 spots and need to implem
> That would only be true if you used random characters in your 17-character
> passphrase. In fact, English text has somewhere between 0.6 and 1.5 bits of
> randomness per letter, whereas an SHA1sum has no more than 4 bits of
> randomness per letter. Let's assume the higher number of randomness f
On Thu, Jul 1, 2010 at 12:53 PM, Tilghman Lesher wrote:
>
> That would only be true if you used random characters in your 17-character
> passphrase. In fact, English text has somewhere between 0.6 and 1.5 bits
> of
> randomness per letter, whereas an SHA1sum has no more than 4 bits of
> randomne
What determines how long SIP channel waits, when you dial a peer with no
registration, before returning ${DIALSTATUS} "CONGESTION"?
When I dial a peer with no registration, SIP channel currently waits
many seconds before returning ${DIALSTATUS} "CONGESTION" - how can I
shorten this timeout?
--
_
On Thursday 01 July 2010 07:43:38 William Stillwell (Lists) wrote:
> Also, technically your "101This is a salt" is stronger than your SHA1 Hash.
>
> Let's say you stick with the "17 character password"
>
> You are using 0-9, a-z, A-Z, and space.
>
> 0-9 = 10
> a-z = 26
> A-Z = 26
> Space = 1
> Tota
On Thu, Jul 1, 2010 at 11:52 AM, wrote:
> Thanks a lot.
>
> Applying the patch gives me a
>
> Hunk #5 failed at 9881
>
>
>
> -Original Message-
> From: Ryan Wagoner
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>
> Sent: Thu, Jul 1, 2010 5:37 pm
> Subject: Re: [asterisk-
On 1 Jul 2010, at 16:56, unsero...@aol.com wrote:
>
> Sorry, i wanted to know what "is in trunk" means.
> So it seems to mean "is in the pipeline for the next version".
DON'T reply to people off list. And stop bloody top posting.
Steve
--
___
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users] Update the LCD with the callee's name after
dialing
On 1 Jul 2010, at 16:25, unsero...@aol.com wrote:
> Sorry, what does this mean? Only in trunk?
If you look in the post you quoted
"This feature is in Asterisk trunk and will be present in the upcoming 1.8
release."
First sentence.
S
--
___
http://svnview.digium.com/svn/asterisk/trunk/
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Jul 1,
On Thu, Jul 1, 2010 at 11:29 AM, wrote:
> Sounds great.
>
> Could you please give me a hint how to install the patch?
> Sorry for my stupid question but I'm a newbie to Asterisk.
>
> Thanks.
>
>
>
> -Original Message-
> From: Ryan Wagoner
> To: Asterisk Users Mailing List - Non-Commercia
Sounds great.
Could you please give me a hint how to install the patch?
Sorry for my stupid question but I'm a newbie to Asterisk.
Thanks.
-Original Message-
From: Ryan Wagoner
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thu, Jul 1, 2010 5:06 pm
Subject: Re
On Thu, Jul 1, 2010 at 9:16 AM, Ishfaq Malik wrote:
> Hi
>
> We've just noticed attempts (close to 20 attempts, sequential peer
> numbers) at guessing peers on 2 of out servers and thought I'd share the
> originating IPs with the list in case anyone wants to firewall them as we
> have done
>
Sorry, what does this mean? Only in trunk?
-Original Message-
From: Steve Howes
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Thu, Jul 1, 2010 5:02 pm
Subject: Re: [asterisk-users] Remote Party ID issue
On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:
> [Jul
Only in trunk...(1.8)
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Jul 1, 2010 at 11:02 AM, Steve
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle wrote:
> Ryan Wagoner wrote:
>>
>> together one for 1.4 that compiles. I'll post both to the list
>> hopefully later today.
>>
>>
>
> Thank you!
>
> Doug
>
> --
>
The Asterisk 1.6.1.20 and 1.4.33.1 patches are almost identical. Both
compile but need to b
On 1 Jul 2010, at 15:52, unsero...@aol.com wrote:
> [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function
> CONNECTEDLINE not registered
> Same happens trying function CALLEDID.
>
> I am using Asterisk 1.6.1.20.
>
> What do i have to do to use this function or alternatively the f
Yes, you are missing a whole bunch of configurations from creating SIP users
to making sure they show as peers on Asterisk to making sure you use dnid,
etc.You probably might want to search google for some configuration help
On Wed, Jun 30, 2010 at 11:24 AM, gokulakrishnan wrote:
> Hi All,
>
Thanks a lot. I will look into it.
On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wrote:
> On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce wrote:
>
>> Thanks a lot.
>>
>> -Bruce
>>
>>
>> On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone wrote:
>>
>>> Hi bruce,
>>>
>>> SIPDefault.conf
>>>
>>>
> I th
Hi,
i have the same problem. Trying to use the dialplan function CONNECTEDLINE()
this way
Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)})
Set(CONNECTEDLINE(num)=${EXTEN})
ends with
[Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function
CONNECTEDL
Hi!
> Has anyone had experience installing it?
> yum install asterisk-chan_misdn
> I'ts the latest Trixbox Distro version and same issues exists if add in
> the Trixbox repo. FAILS as per below
Please search this list for recent messages on mISDN, or Google it.
You will find that mISDN v1 doe
On Wed, 30 Jun 2010, Steve Edwards wrote:
>> Now I whipped up a C program to create a call file to do the same thing
>> from the command line:
>>
>> [snip]
>> fprintf(callfile, "Channel: Local/*...@custom-callfwd/n\n");
>
> I don't see exten "*71" in custom-callfwd.
Doh! That was the pr
Hi
We've just noticed attempts (close to 20 attempts, sequential peer
numbers) at guessing peers on 2 of out servers and thought I'd share the
originating IPs with the list in case anyone wants to firewall them as
we have done
109.170.106.59
112.142.55.18
124.157.161.67
Ish
--
Ishfaq Ma
Hi,
I am using CEL to more accurate billing information with some success. However
there is an ambiguity in the CEL data when multiple destinations are specified
in the DIAL command.
For example, if I have
Dial(SIP/outboundA/100&SIP/outboundA/101&SIP/outboundB/200&SIP/outboundB/201)
this is r
On Thu, 1 Jul 2010 15:26:27 +0300, Tzafrir Cohen
wrote:
>Re-run ./configure
Ah, hadn't thought of this :-/
>The Debian asterisk package depends on liblua5.1-0-dev and builds
>pbx_lua just fine.
Yes, it did compile after re-running ./configure, make menuconfig,
make.
I'll check how to use exten
Hi,
Has anyone had experience installing it?
yum install asterisk-chan_misdn
I'ts the latest Trixbox Distro version and same issues exists if add in the
Trixbox repo.
FAILS as per below:
I have a ISDN single port PCI BRI card installed and detected.
__
Loaded plugins: fastestmirr
Also, technically your "101This is a salt" is stronger than your SHA1 Hash.
Let's say you stick with the "17 character password"
You are using 0-9, a-z, A-Z, and space.
0-9 = 10
a-z = 26
A-Z = 26
Space = 1
Total Possible Values = 63
17^63 = 3.2982384238829760312713680399948e+77
Your sha1 is
Ryan Wagoner wrote:
>
> together one for 1.4 that compiles. I'll post both to the list
> hopefully later today.
>
>
Thank you!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
--
Unfortunately not. I did it a few times using a php script using a 'which'
loop to create multiple call files. You can also do it in a dialplan which
is a slow process. I have it described at:
http://ilovetovoip.com/2010/03/calling-multiple-extensions-and-let-them-all-answer/
Zeeshan A Zakaria
-
On Thu, Jul 1, 2010 at 7:49 AM, Deepesh D wrote:
> So that both extensions 101 and 102 rings simultaneously.
>
Yes, or use a local channel to dial multiple extensions.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeaco
On Thu, Jul 01, 2010 at 01:21:31PM +0200, Gilles wrote:
> On Thu, 01 Jul 2010 01:32:08 +0200, Gilles
> wrote:
> >I'm taking a look at how to write scripts to be called from the
> >dialplan, and saw pbx_lua mentioned.
>
> I'm not having much luck adding the pbx_lua module to Asterisk (on a
> Ubunt
On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle wrote:
> CunningPike wrote:
>> On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell wrote:
>>
>>
>> We use the patch in https://issues.asterisk.org/view.php?id=6643. Works
>> great.
>>
>>
>
> There is a much newer patch for 1.4 that can be found at:
>
> https:
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101&SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
--
___
On Thu, 01 Jul 2010 01:32:08 +0200, Gilles
wrote:
>I'm taking a look at how to write scripts to be called from the
>dialplan, and saw pbx_lua mentioned.
I'm not having much luck adding the pbx_lua module to Asterisk (on a
Ubuntu 10.04) :-/
# apt-get install lua5.1 liblua5.1-0 liblua5.1-0-dev
#
Hi!
> The 'M' in PtMP stands for 'Multi'. Basically PTP is the standard ISDN
> protocol, and PtMP is an extension of its logic to make ISDN (BRI) phones
> behave somewhat like analog phones: allow you to connect several of them
> on the same line.
In other words:
While you *must* have exactly on
On Thu, 01 Jul 2010 15:22:33 +0500, Faisal Hanif
wrote:
>I am in process of merging all my AGIs+Dialplan to a single LUA
>dialplan. It seems much interesting to me spacial LUA tables which allow
>me to support a complete object like programming. Yet I did not
>completed / tested.
Thanks for th
Hi,
I am in process of merging all my AGIs+Dialplan to a single LUA
dialplan. It seems much interesting to me spacial LUA tables which allow
me to support a complete object like programming. Yet I did not
completed / tested.
Regards,
*Faisal Hanif*On 7/1/2010 2:37 PM, Gilles wrote:
On We
Thank you for your replies.
@Olivier
Yes. The telco here (in Mumbai, India) charges more for p2p. See their
tariff http://mtnlmumbai.in/telecomservices/isdntariff.html#bratariff
It is mentioned in "Charges for point to point connectivity : ISDN BRA
Lines" that extra charges would be applied for p
On Wed, Jun 30, 2010 at 05:56:27PM -0500, Alex Villacís Lasso wrote:
> I have reproduced this stream of warnings on another machine with
> asterisk-1.4.33.1 and dahdi-2.3.0.1, and also with other card types
> (OpenVox with 1 E1 port, Sangoma with 2 T1 ports, Rhino with 2 T1
> ports), so I do n
CunningPike wrote:
> On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell wrote:
>
>
> We use the patch in https://issues.asterisk.org/view.php?id=6643. Works great.
>
>
There is a much newer patch for 1.4 that can be found at:
https://issues.asterisk.org/view.php?id=8824
But, it won't apply c
On Thu, Jul 01, 2010 at 01:21:23PM +0530, pranav jawale wrote:
> Hello,
>
> I'm a graduate student. We are setting up an IVR system for research purpose
> on a BRI channel. (We can't afford PRI line as its cost is about 10x of the
> BRI). The line will be connected to the CTI card. Using asterisk
On Wed, Jun 30, 2010 at 11:50:49PM -0500, Tilghman Lesher wrote:
> On Wednesday 30 June 2010 18:38:51 Steve Edwards wrote:
> > On Sun, 13 Jun 2010, Tilghman Lesher wrote:
> > > I would generally suggest something a little more deterministic (where
> > > 101 is your extension):
> > >
> > > $ echo '1
On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
wrote:
>I've never used it (I'm a 1.2 Luddite), but I would be very interested in
>anything that looks like a "real" language for writing dialplans.
That's why I'm interested in using Lua to write dialplan scripts,
besides the fact that due
"William Stillwell (Lists)" writes:
> I have several remote phones that experience a slight call delay when
> answering phones, ie, they will answer, speak a few words, and then the
> remote caller will hear them, and the first half is cutoff?
This is actually a somewhat common problem in SIP
Hello,
2010/7/1 pranav jawale
> Hello,
>
> I'm a graduate student. We are setting up an IVR system for research
> purpose on a BRI channel. (We can't afford PRI line as its cost is about 10x
> of the BRI). The line will be connected to the CTI card. Using asterisk
> server we will be recording t
Hello,
I'm a graduate student. We are setting up an IVR system for research purpose
on a BRI channel. (We can't afford PRI line as its cost is about 10x of the
BRI). The line will be connected to the CTI card. Using asterisk server we
will be recording the calls.
I'm confused about whether we sho
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.
>>>
>>> -Matt
>>>
>>> On Mon, Jun 28, 2010 at 3:48
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 Latham wrote:
>>> Remote Party ID in trunk, it work
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