On 5/27/11 6:33 PM, Gordon Henderson wrote:
Personally I'd avoid Patton. No-one has a clue how to configure them.
I've struggled for the past couple of days and have given up and they're
being sent back to be replaced by Mediatrix boxes.
Then you're asking the wrong people. It is totally
I read about asterisk 1.10 in website https://wiki.asterisk.org. but
didnt find this release from asterisk community.
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On 1 Jun 2011, at 22:50, Jesse Thompson wrote:
We are managing an Asterisk installation for residential VOIP service, and we
are having a problem where all inbound calls to DIDs which are assigned to us
by our wholesaler but not yet assigned to a downstream customer get caught in
a routing
See this link for release date...
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
[SATISH]
On Thu, Jun 2, 2011 at 1:09 PM, Nikhil d.nik...@cem-solutions.net wrote:
I read about asterisk 1.10 in website https://wiki.asterisk.org. but didnt
find this release from asterisk
OK thanks a lot for your help now all is ok :)
2011/5/31 salaheddine elharit salah.elharit...@gmail.com
Hello
after remove the _ and put the number like that 0678922645,1, the issue has
been solved
thank you so much :)
2011/5/31 mahesh katta maheshka...@flexydial.com
Remove the _
Nikhil,
This is how I would implement '3 way conference' in Asterisk with the help
of dynamic features.
Assume 3 SIP friends 1110, and 1112 in sip.conf. For 1110 in sip.conf,
context=test3way
Add following in applicationmap section of features.conf
[applicationmap]
3way-start =
Hi everyone,
i ended using Patton, right after a fiasco with overlap to enblock
conversion
on a users PBX ...
Had to go out and pay more for a Voxip unit . That things serves me well
and is
far less complicated to setup.
NOw using it to interconnect a PBX with aprox 5000 extensions behind
We use Jira at work. I hate it. Hope you have a better experience than
I've had!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell
Bryant
Sent: Wednesday, June 01, 2011 7:51 PM
To:
can someone explain to me the benefits of upgrading to version 1.8?
we are currently running 1.6
I know one benefit of 1.8 is digium supports it
also, how stable is version 1.8 compared to 1.6? Thank you for you input.
--
_
--
1.8 is stable when compared to 1.6, also in 1.8 you will get Long Term
support.
On Thu, Jun 2, 2011 at 6:31 PM, vip killa vipki...@gmail.com wrote:
can someone explain to me the benefits of upgrading to version 1.8?
we are currently running 1.6
I know one benefit of 1.8 is digium supports it
So many new features have been added in 1.8.
Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8
Nope, Asterisk 1.8 is not stable enough yet.
[SATISH]
On Thu, Jun 2, 2011 at 6:33 PM, Gopal krishnan
gopalakrishnan...@gmail.comwrote:
1.8 is stable when compared to 1.6, also in
what do you mean Asterisk 1.8 is not stable enough yet? Can you give
specific examples/scenarios?
On Thu, Jun 2, 2011 at 9:28 AM, Satish Barot satish4aster...@gmail.comwrote:
So many new features have been added in 1.8.
Check this...https://wiki.asterisk.org/wiki/display/AST/New+in+1.8
Nope,
Hi Does anyone know of an accurate resource I could refer to for this?
The best I can find is
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
And that table wont create in my database...
Thanks
Ish
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
--
On 01/06/11 16:13, Allen David Niven wrote:
what does ossec give u that fail2ban does not ?
thx and cheers
Replied to list so others can find this in the future if they want to.
I haven't spent a lot of time investigating fail2ban as I was already
using ossec before I saw much talk about
To clarify; I observe the exact same results no matter how I connect
to the AMI on this particular server. I tried connecting FROM this
server to an AMI on another server to make sure it wasn't the telnet
client or some such, and then it worked perfectly.
To answer the question, if I use the
On 11-06-02 09:35 AM, vip killa wrote:
what do you mean Asterisk 1.8 is not stable enough yet? Can you give
specific examples/scenarios?
I too would like to see a specific example, additionally if you can
create an test using the testsuite I'll be happy to review it and merge
the code into
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Paul Belanger
Sent: Thursday, June 02, 2011 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] benefits of asterisk 1.8
On
2011/6/2 Örn Arnarson o...@arnarson.net:
To clarify; I observe the exact same results no matter how I connect
to the AMI on this particular server. I tried connecting FROM this
server to an AMI on another server to make sure it wasn't the telnet
client or some such, and then it worked
On 06/02/2011 06:46 AM, Terry Brummell wrote:
We use Jira at work. I hate it. Hope you have a better experience than
I've had!
We've been using it for years internally to Digium. We've been happy
with it.
--
Russell Bryant
Digium, Inc. | Engineering Manager, Open Source Software
445
On 06/02/2011 10:29 AM, Eric Wieling wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Paul Belanger
Sent: Thursday, June 02, 2011 11:27 AM
To: asterisk-users@lists.digium.com
Subject: Re:
On Thu, 2 Jun 2011, Andreas Sikkema wrote:
On 5/27/11 6:33 PM, Gordon Henderson wrote:
Personally I'd avoid Patton. No-one has a clue how to configure them.
I've struggled for the past couple of days and have given up and they're
being sent back to be replaced by Mediatrix boxes.
Then
Well,
About sipvicious, just put a kamailio in front of asterisk and just drop
all messages with user agents corrreponding to these messages.
Spivicious first send options messages, read the user agent and drop if
it's corresponding to one of the user agents well known to be used.
In
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
On Wed, Jun 1, 2011 at 12:22 PM, Ira i...@extrasensory.com wrote:
At 10:56 AM 6/1/2011, you wrote:
Do you have:
sip.conf
[general]
allowguest=no
So
If you can use curl, and can do some text parsing and know regular
expressions, you may be able to use this free CNAM service:
http://www.numberguru.com/ and integrate into your system. This one appears
to have a more complete database. When I tried my number, I have gotten my
full name, but
Hi Guys,
Actually My question is as in the subject, may I use a regular phone line to
receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8.
--
Khalid
--
_
-- Bandwidth and Colocation Provided by
I just checked several of my numbers and several others known to me, it
really isn't much better
2 of them returned names other than mine, and all had the wrong city,
though at least the state was correct.
All but one also had the wrong carrier.
I fear these databases are are so full of errors
Also you guys may need to use:
sip.conf
[general]
allowguest=no
*alwaysauthreject = yes*
On Thu, Jun 2, 2011 at 1:01 PM, Al lists asteris...@gmail.com wrote:
I'll check this option and see if it helps next time,
just to clarify, there were no actual calls in place, just DOS register
attack.
Good afternoon,
I'm trying to write a simple callback context, but i need to hangup an
incoming call and then call the origin number back, the problem is that
asterisk stops processing the call after Hangup() application then it is
not able to dial the origin number back.
Sorry for the
The Asterisk Development Team has announced the release of Asterisk
version 1.8.4.2, which is a security release for Asterisk 1.8.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 1.8.4.2 resolves an issue
en_US.UTF-8 in all cases.
On Thu, Jun 2, 2011 at 3:33 PM, Mark Deneen mden...@gmail.com wrote:
2011/6/2 Örn Arnarson o...@arnarson.net:
To clarify; I observe the exact same results no matter how I connect
to the AMI on this particular server. I tried connecting FROM this
server to an AMI on
Letting a carrier use you as a carrier seems like quite a bad idea generally..
I think I would agree. :)
_NXXNXX = Dial(SIP/${EXTEN}@upstream,120); // numbers not handled here
get routed upstream
in the 'local' context instead of the other one?
So here is where the finer
Is this available in current SVN ?
Date: Thu, 2 Jun 2011 15:07:50 -0400
From: asteriskt...@digium.com
To: asteriskt...@digium.com
Subject: [asterisk-users] Asterisk 1.8.4.2 Now Available (Security Release)
The Asterisk Development Team has announced the release of Asterisk
version
Hi Guys!
If i reload my asterisk it create /var/log/asterisk/* file with root
permission. I am running asterisk with asterisk user and group. Do you have
any idea ?
root@campbx1:~# ls -l /var/log/asterisk/
total 716
drwxr-xr-x 2 asterisk asterisk 4096 2011-05-06 15:38 cdr-csv
drwxr-xr-x 2
Hi all,
Let's get some feedback going here and see if there is any general support
in a user-driven CNAM concept.
Assuming that your landline/mobile outbound provider does not push
caller-name + number for you with your calling plan. Would you pay $1/yr to
have the access to update your
2011/6/2 Antonio Modesto mode...@isimples.com.br
Good afternoon,
I'm trying to write a simple callback context, but i need to hangup an
incoming call and then call the origin number back, the problem is that
asterisk stops processing the call after Hangup() application then it is not
able
On 02/06/11 03:35 PM, satish patel wrote:
Is this available in current SVN ?
Changes are always checked into SVN first and then made available in a tag.
Leif.
--
Leif Madsen
http://www.oreilly.com/catalog/asterisk
--
_
--
Hello,
I am implementing a small ACD system on Asterisk 1.6.2.17.2 I need help with
ChannelRedirect. I have a caller connected to an agent. The agent may request
additional help by consulting another department. I can't use manual process
with blind or directed transfer as the agent have many
On 5/29/2011 8:55 AM, Richard Kenner wrote:
What happens when the CNAM is changed? How often does it go back and poll
the database?
That's actually a very very good question! Are entries in the database
given a TTL/Expiration before being checked/researched again?
Slainte,
Sherwood
Cool topic!
Our company (generationD) developed some CID scripts for free use, and we would
be interested in building and hosting this service.
On the spec side, how do we avoid users claiming numbers belonging to others?
(Could be an admin nightmare)
Do we allow number ranges?
Do we require
Hello,
I am searching for a DTMF issue on my setup ( 2 years and counting ),
and I am wondering why rtp.c has code to mute DTMF ( the rtp-dtmfmute
variable ), but this same mechanism does not exist in dahdi.
I am sending a DTMF over SIP w/ RTP RFC2833 to the asterisk box with
the dahdi
Paul,
With due respect to Digium work, are there no issues with Asterisk 1.8?
https://issues.asterisk.org/view_all_bug_page.php
[SATISH]
On Thu, Jun 2, 2011 at 9:21 PM, Kevin P. Fleming kpflem...@digium.comwrote:
On 06/02/2011 10:29 AM, Eric Wieling wrote:
-Original Message-
Use Asterisk Application 'System()' in h extension to
create callfile which will handle your callback.
You can also try for 'Originate()' application.
[SATISH]
2011/6/3 Antonio Modesto mode...@isimples.com.br
Good afternoon,
I'm trying to write a simple callback context, but i need to
Warren,
A good example given.
Just suggest to use 'Move' instead of 'Copy' for placing callfile in
outgoing folder.
A J Stiles has explained it in a better way in one of his replies.
http://lists.digium.com/pipermail/asterisk-users/2011-May/262929.html
[SATISH]
On Fri, Jun 3, 2011 at 1:16 AM,
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