On Wed, Jun 30, 2010 at 12:10 PM, CunningPike cunningp...@gmail.com wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com 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 lath...@gmail.com
On Wed, Jun 30, 2010 at 4:26 PM, Ryan Wagoner rswago...@gmail.com wrote:
On Wed, Jun 30, 2010 at 6:10 PM, CunningPike cunningp...@gmail.com wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnell mattdarn...@gmail.com wrote:
Thank you Andrew,
I will check it out. We are currently running 1.4.
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
Hello,
2010/7/1 pranav jawale pranavshri...@gmail.com
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
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.
On Wed, 30 Jun 2010 17:01:50 -0700 (PDT), Steve Edwards
asterisk@sedwards.com 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,
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 '101This is a
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
CunningPike wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com 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,
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 not
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
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
On Thu, 01 Jul 2010 15:22:33 +0500, Faisal Hanif fai...@vopium.com
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.
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 one
On Thu, 01 Jul 2010 01:32:08 +0200, Gilles codecompl...@free.fr
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
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
--
On Thu, Jul 1, 2010 at 5:55 AM, Doug Lytle supp...@drdos.info wrote:
CunningPike wrote:
On Tue, Jun 29, 2010 at 2:53 PM, Matt Darnellmattdarn...@gmail.com 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
On Thu, Jul 01, 2010 at 01:21:31PM +0200, Gilles wrote:
On Thu, 01 Jul 2010 01:32:08 +0200, Gilles codecompl...@free.fr
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 Thu, Jul 1, 2010 at 7:49 AM, Deepesh D deep.d2...@gmail.com 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
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
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.
--
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
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:
On Thu, 1 Jul 2010 15:26:27 +0300, Tzafrir Cohen
tzafrir.co...@xorcom.com 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
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/100SIP/outboundA/101SIP/outboundB/200SIP/outboundB/201)
this is
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
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 problem. In
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 does
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
Thanks a lot. I will look into it.
On Wed, Jun 30, 2010 at 11:15 AM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Jun 30, 2010 at 8:40 AM, bruce bruce bruceb...@gmail.com wrote:
Thanks a lot.
-Bruce
On Wed, Jun 30, 2010 at 4:55 AM, Emanuele Carbone carbe...@gmail.comwrote:
Hi
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
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
On Thu, Jul 1, 2010 at 8:41 AM, Doug Lytle supp...@drdos.info 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
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
Sorry, what does this mean? Only in trunk?
-Original Message-
From: Steve Howes steve-li...@geekinter.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:02 pm
Subject: Re: [asterisk-users] Remote Party ID issue
On Thu, Jul 1, 2010 at 9:16 AM, Ishfaq Malik i...@pack-net.co.uk 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
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 rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Thu, Jul 1, 2010 at 11:29 AM, unsero...@aol.com 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 rswago...@gmail.com
To: Asterisk Users
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 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
--
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thu, Jul 1, 2010 5:37 pm
Subject: Re: [asterisk-users]
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
--
On Thu, Jul 1, 2010 at 11:52 AM, unsero...@aol.com wrote:
Thanks a lot.
Applying the patch gives me a
Hunk #5 failed at 9881
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
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
Total Possible
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 Thu, Jul 1, 2010 at 12:53 PM, Tilghman Lesher tles...@digium.com 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
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 for
-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
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:
-Original Message-
From: Ryan Wagoner rswago...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
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,
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.
--
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:
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
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
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
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 personally
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 tles...@digium.com wrote:
On
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
-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
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 -
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
On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
pretty much giving up on Skype for Asterisk (and Skype for SIP) now
that I realize that they'll be
On Thu, Jul 1, 2010 at 3:50 PM, Kyle Kienapfel doctor.w...@gmail.com wrote:
On Wed, Jun 16, 2010 at 9:23 AM, Jeff LaCoursiere j...@sunfone.com wrote:
On Wed, 16 Jun 2010, Randy R wrote:
On Wed, Jun 16, 2010 at 5:16 PM, Jeff LaCoursiere j...@sunfone.com
wrote:
pretty much giving up on Skype
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.
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
--
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-- Bandwidth and
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=230492577678#ht_500wt_1076
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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-pilot
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
Which
On Thu, Jul 1, 2010 at 7:09 PM, Miguel Molina mmol...@millenium.com.cowrote:
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
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 Fri, Jul 2, 2010 at 1:07 AM, Kyle Kienapfel doctor.w...@gmail.com 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
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
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
I want
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