On Mon, Jul 27, 2009 at 01:28:59PM -0300, Gustavo A Gonzalez wrote:
Hello all! Im running asterisk 1.4.23 and sometimes it crashes. Because I
need to look for what asterisk crashes I run asterisk with option '-g' for
debugging purpose. When I search for core files in filesystem nothing
First off, you should post a new message rather than replying to an
existing message.
On Tue, Jul 28, 2009 at 08:18:42PM -0400, John F. Ervin wrote:
Never having actually rolled an Asterisk (Trixbox in my case) system
into production. I was wondering if in most peoples opinion if given
hello,
I have SIP phone registered with my server
now if they send me any number for dialing then i want to give a response
code
actually this number is conference number and i need to chek via DB query
that this conference is valid or not
if conference is not valid then i want to send a
On Tue, 28 Jul 2009, John F. Ervin wrote:
Never having actually rolled an Asterisk (Trixbox in my case) system
into production. I was wondering if in most peoples opinion if given
the choice would rather have a straight VOIP/SIP system or would rather
have a system with normal POTS/analog
So, there's two kinds of authentication that routinely go on in the SIP
client/server world:
1) REGISTER authentication -- this is the 401 Unauthorized challenge to
an initial REGISTER request that causes it to be resent with
WWW-Authorize headers containing various authentication credentials,
An application commanding asterisk with AMI is going to launch lots of
concurrent calls in very few seconds using the Originate AMI command but
it's also going to need to be able to cancel very quickly any call of them
even before each OriginateResponse event comes in. All the calls will be
done
When you removed the mailboxes, you either messed up permissions or just
made the files unavailable to Asterisk. You presumably still have
/var/spool/asterisk/voicemail/default, so you need to check that tree vs the
tree on a working server.
_
From:
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
Hi,
over the year, the IM question popped up here and then, the answer was
allways something along the line yeah, there's an experimental patch, but
it's not yet ready...
Are there any plans to change this, to ie. allow messages between xmpp and
sip or iax? Especially with the chan_skype (that's
Hi -
I had a client recently move their asterisk system (asterisk 1.4.26,
dahdi 2.2.0.1, aex800 w/vpm module) to a new location, a building
that's nearly 150 years old. I was not personally able to go there,
but the person who did the move said the building's demarc room was
scary-- water leaks,
Steve Totaro escribió:
On Tue, Jul 28, 2009 at 9:13 PM, Miguel Molina
mmol...@millenium.com.co mailto:mmol...@millenium.com.co wrote:
John F. Ervin escribió:
Never having actually rolled an Asterisk (Trixbox in my case) system
into production. I was wondering if in most
Jared Smith wrote:
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
Alex Balashov wrote:
I wouldn't approach this by trying to rework the CDRs at all; CDRs are
fundamentally low-level call records. They correspond to calls.
If you need logic to support a billing model for some specific
application (i.e. time after connect to agent), I would approach that
Thank you guys, for this clarification...
Arturo Ochoa
Electrosystems
2009/7/28 Kevin P. Fleming kpflem...@digium.com
Miguel Molina wrote:
Counting that everything works well on the IP portion of the
communication, you might have something, but the store and forward
process that has
Noah Miller wrote:
My question for anyone with knowledge on this: would HPEC do a better
job than the VPM module (or oslec)? Can HPEC cope with very long echo
tails?
HPEC and the Digium VPMADT032 use the same algorithms from the same vendor.
--
Kevin P. Fleming
Digium, Inc. | Director of
Jose Arias cyr2...@gmail.com writes:
An application commanding asterisk with AMI is going to launch lots of
concurrent calls in very few seconds using the Originate AMI command but it's
also going to need to be able to cancel very quickly any call of them even
before each OriginateResponse
My question for anyone with knowledge on this: would HPEC do a better
job than the VPM module (or oslec)? Can HPEC cope with very long echo
tails?
HPEC and the Digium VPMADT032 use the same algorithms from the same vendor.
Aha. Thanks for this tidbit, Kevin!
Next question: does anybody
Noah Miller wrote:
Next question: does anybody know how to handle extremely long tail
echo that a VPM module cannot?
How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can
handle 128ms echo tails, which is pretty darn long. It's rare to see an
echo tail longer than that except on
Greetings to all.
this is my first question, and but that nothing is for consulting if this with
asterisk can be realised. I have a commutator 3com, connected to 20 telephones
of the same mark, my necessity right now is to be able to record the calls that
enter the commutator, and wanted to
if you are running Asterisk in front of the other pbx you can record the
calls that you send to the other system.
You will either need some type of pri interface to connect between the two
systems if digital and some fxs interfaces if analog.
Tom
_
From:
Check OrecX out. The non GPL version may be able to do NBX protocol.
Not sure what a 3com commutator unless that is what Digium and 3com teamed
up on.
Thanks,
Steve
On Wed, Jul 29, 2009 at 3:35 PM, Tom Moore tommym2...@gmail.com wrote:
if you are running Asterisk in front of the other pbx
Next question: does anybody know how to handle extremely long tail
echo that a VPM module cannot?
How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can
handle 128ms echo tails, which is pretty darn long. It's rare to see an
echo tail longer than that except on very high latency
They excuse my question again.
I explain to them.
I have an equipment 3com v3000 with 20 extensions, my question is if I can use
asterisk to record the calls,
is necessary an additional servant? at the moment only I have the telephones
connected to this equipment. Where encounter
Mr. Rodriguez wrote:
They excuse my question again.
I explain to them.
I have an equipment 3com v3000 with 20 extensions, my question is if I
can use asterisk to record the calls,
is necessary an additional servant? at the moment only I have the
telephones connected to this equipment.
Some times i used babelfish :(
Carlos Rodriguez
Date: Wed, 29 Jul 2009 17:39:33 -0400
From: sean.bri...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Recording calls again
Mr. Rodriguez wrote:
They excuse my question again.
I explain to them.
Each year at AstriCon, we have an Open Source Pavilion which showcases
projects which are adjuncts to Asterisk, or which are directly
relevant to improving the utility and features of Asterisk. It gives
smaller projects the chance to have some space to display what they're
doing, and to
Jared Smith wrote:
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
deny=0.0.0.0/0.0.0.0
Anthony wrote:
Jared Smith wrote:
On Tue, 2009-07-28 at 16:06 -0700, Bruce Ferrell wrote:
I have a carrier who tells me he will be sending me traffic from a wide
range of IP addresses.
so I set up a realtime peer as follows:
[peer]
defaultip=xxx.xxx.xxx.xxx
host=xxx.xxx.xxx.xxx
Noah Miller wrote:
Next question: does anybody know how to handle extremely long tail
echo that a VPM module cannot?
How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can
handle 128ms echo tails, which is pretty darn long. It's rare to see an
echo tail longer than that
On Wed, Jul 29, 2009 at 5:25 PM, Mr. Rodriguezclubtorr...@hotmail.com wrote:
They excuse my question again.
I explain to them.
I have an equipment 3com v3000 with 20 extensions, my question is if I can
use asterisk to record the calls,
is necessary an additional servant? at the moment only I
In the last couple of years I can only think of two sites where we've installed
E1 connections, compared to many tens of sites where we've gone with VoIP only.
You can mitigate against QoS issues by simply installing two DSL connections
(one for internet, one for voice). With a decent load
Noah Miller wrote:
Next question: does anybody know how to handle extremely long tail
echo that a VPM module cannot?
How long is 'long' in this case? The VPMs and HPEC (and OSLEC) can
handle 128ms echo tails, which is pretty darn long. It's rare to see an
echo tail longer than that
On Tue, Jul 28, 2009 at 1:01 AM, hadi motamedimotamed...@gmail.com wrote:
Dear All
Can you please let us know how we can modify our Asterisk inter digit delay
? Actually , our subs dials his intended numbers with some delay in between
entering the digits sequentially . It seems that our
John Todd wrote:
What your project should have:
- No significant corporate sponsorship
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083
On Sun, Jul 26, 2009 at 1:19 PM, Chris
Douglaschris.doug...@pioneerballoon.com wrote:
We have an inbound PRI connected to our Cisco 3825 router which is then
passing the calls to Asterisk as SIP calls. We're getting the CallerID
number but not the CallerID name. We are seeing the name in the
I'm pretty new to this whole Asterisk system VoIP thing, but being a
programmer by trade the complexity didn't scare me off (at least not yet)...
I have setup an Asterisk system for my home home office. My wife I
run two separate businesses from home, and we have a general family home
On Wed, 29 Jul 2009, Myles Wakeham wrote:
I'm pretty new to this whole Asterisk system VoIP thing, but being a
programmer by trade the complexity didn't scare me off (at least not yet)...
I have setup an Asterisk system for my home home office. My wife I
run two separate businesses from
Not when you consider that there are plenty of spaces for corporate projects
as well.
On Wed, Jul 29, 2009 at 9:21 PM, Anthony antho...@rockynet.com wrote:
John Todd wrote:
What your project should have:
- No significant corporate sponsorship
JT
---
John Todd
On Wed, 29 Jul 2009, Myles Wakeham wrote:
I have setup an Asterisk system for my home home office.
[snip]
The cost of all these lines with analog carriers was getting ridiculous,
so I'm moving over to a SIP carrier. I created one account for a single
phone number with a SIP carrier
Are there any other phones registered, or is it just this phone that is
having issues? The first thing that I see is the qualify=200 line, and I
have not had good experience with Cisco devices and any qualify setting. I
would try leaving that out. I also have double quotes around the line1_*
Dear David
I mean when the subs dials the digits with some delay between entering the
digits sequentially . At our current case , the Asterisk will wait about 2
seconds to see if another digit will be dialed or not and then he will route
the dialed digits according to the pre-defined routing table
Thank-you for your email. I will be out of the office from Thursday, July 30
until Monday, August 3. I will have limited access to email during this time
and will respond as soon as possible.
If you need immediate assistance, please call our support line at 770-674-3900
x 1.
Thank you.
I mean when the subs dials the digits with some delay between entering the
digits sequentially . At our current case , the Asterisk will wait about 2
seconds to see if another digit will be dialed or not and then he will route
the dialed digits according to the pre-defined routing table or he
Thank you very much for your reply . But please be informed that our current
line-outgoing route is being configured as the followings (in
extensions.conf):
[line-outgoing]
exten = _X.,1,macro(dialuser,Zap/g1/${EXTEN},${EXTEN})
As you see , it is trying to consider the dialed number as an
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