Good morning,
from the last question i assume you're looking for a SIP-based
configureation.
On 03-30-2011 00:16, bilal ghayyad wrote:
1) How I can assign for each button an extension?
>
you can configure them as lines (at least in my 7940). look for
linex_name, linex_authname and linex_p
Hi,
I'm using IAX2 between our SIP and PSTN servers, both running Asterisk
1.6.2. Users connect to the SIP server and dial; the SIP server
forwards the call to the PSTN server over IAX2, which then dials out
over the connected PRI. Since users need detailed call progress
feedback, the first
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On Tue, Mar 29, 2011 at 4:03 PM, Warren Selby wrote:
> That information does indeed look like what I want and it appears to be
> setup correctly. I will be building a comparable test system later today
> (using all the same software versions as you) and I'll test to see if I get
> the same issue
The answer to all of your questions are the same - the config file that you
create for your phone.
Thanks,
--Warren Selby, dCAP
On Mar 29, 2011, at 5:16 PM, bilal ghayyad wrote:
> Hello;
>
> I need to use Cisco IP Phones with Asterisk and I have some questions to know
> how to use it if som
Sorry for the top post, responding from my phone...
Yes, you'll need both DAHDI and libpri to make an E1 card work with asterisk.
Yes, you'll most likely use the Dial() command inside extensions.conf in order
to dial out. You'll differentiate your PRI channels from your analog channels
in your
Hi Gilles,
Just to provide an alternative to sshguard: you could use BFD[1]
(based on bash scripts) and configure it to use iptables to block the
attacker host.
The default configuration is to check the logs at each 3 minutes
(using a crontab entry).
BFD rules for Asterisk could be found here [2]
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya,
China, Iran, etc?
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
after reviewing last week's log i'd say around 25-28k/min :)
On Tue, 29 Mar 2011, Gilles wrote:
So it l
Hello;
I need to use Cisco IP Phones with Asterisk and I have some questions to know
how to use it if someone can advise:
1) How I can assign for each button an extension?
2) How I can assign for specific button a feature to be used (like call forward
or call pickup .. etc)?
3) As you know that
Hi All;
I have an E1 card with two ports for ISDN PRI.
Do I need to install DAHDI in addition to LIBPRI?
For placing outside calls (outgoing) via the PRI, then in the extension.exe
file, I will use the Dial function? But how can I determine that I need to use
the PRI channels and not the analo
On Tue, 29 Mar 2011 23:09:06 +0200, ad...@3a.hu wrote:
>On 03-29-2011 19:25, Steve Edwards wrote:
>> Really? How many callers are you expecting from North Korea, Libya, China,
>> Iran, etc?
>after reviewing last week's log i'd say around 25-28k/min :)
So it looks like I should check out sshguard i
On 03-29-2011 19:25, Steve Edwards wrote:
Really? How many callers are you expecting from North Korea, Libya, China,
Iran, etc?
after reviewing last week's log i'd say around 25-28k/min :)
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On Tue, Mar 29, 2011 at 3:52 PM, Eric W. Davenport
wrote:
> Hi Warren,
>
> Thanks for your help,
>
> I think this is what you want
>
>
Please don't mail me (or anyone else) off the list directly without being
specifically asked to. The idea is to keep things on the mailing list in
order to help
On Tue, Mar 29, 2011 at 3:57 PM, Cary Fitch wrote:
> Obviously, the other side of the world wants connections to your side, no
> matter what side you are on.
> :-)
>
> Cary
>
>
Exactly
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Obviously, the other side of the world wants connections to your side, no
matter what side you are on.
:-)
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, March 29, 201
Le 29/03/2011 19:34, Sherwood McGowan a écrit :
On 3/29/2011 12:25 PM, Steve Edwards wrote:
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On Tu
Oh, damn, my bad, I've apparently read too many sip.conf entries today
--
Sherwood McGowan
Carrier, ITSP, Call Center, and PBX Solutions Consultant
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New
On 3/29/2011 2:29 PM, Warren Selby wrote:
It looks like you did to me. Is it just OPTIONS packets that are showing
the wrong fromuser field? In other words, when you send call traffic over
this peer, does it properly create the SIP packets? For some reason, I'm
correct - when i actually invi
On Tue, Mar 29, 2011 at 1:25 PM, Jeremy Kister wrote:
> On 3/29/2011 1:56 PM, Sherwood McGowan wrote:
>
>> [mypeer](peer)
>>> host=10.0.138.226
>>> defaultuser=211941
>>> fromuser=211941
>>> md5secret=023f30a320a5781e8ffd1af9888012af
>>> incominglimit=10
>>>
>>
> IIRC, you need to
On 3/29/2011 1:56 PM, Sherwood McGowan wrote:
[mypeer](peer)
host=10.0.138.226
defaultuser=211941
fromuser=211941
md5secret=023f30a320a5781e8ffd1af9888012af
incominglimit=10
IIRC, you need to define the fromuser in the peer in order for the
qualify checks (options packets) to con
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
wrote:
Remember guys, there's a LOT of IP blocks out there that are almost
definitely not going to be somewhere you expect to receive SIP traffic
from.
On Tue, 29 Mar 2011, Gilles wrote:
I agree. Is there a list I could use to check which
On Tue, Mar 29, 2011 at 12:32 PM, Eric W. Davenport
wrote:
> I restarted Asterisk and I still have all the fields outgoing and incoming
> except for the CLID field.
> Clid is populated in the CSV Simple file as well as the CSV Custom file.
>
>
Please, if you would, copy and paste the output of a '
On 3/29/2011 12:52 PM, Jeremy Kister wrote:
> I recently configured a SIP peer which i must specify my fromuser as
> my ten digit "DID". I send calls to this peer, but whenever Asterisk
> sends an options message, the fromuser is "asterisk".
>
> Is this a bug? Or is there some other config I mu
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 211941:123456@10.0.138.2
On 3/29/2011 12:42 PM, Gilles wrote:
> On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
> wrote:
>> Remember guys, there's a LOT of IP blocks out there that are almost
>> definitely not going to be somewhere you expect to receive SIP traffic
>> from.
> I agree. Is there a list I could use to
On Tue, Mar 29, 2011 at 2:34 PM, Sherwood McGowan
wrote:
> On 3/29/2011 12:25 PM, Steve Edwards wrote:
>>> On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
>>
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table
On Tue, 29 Mar 2011 12:34:04 -0500, Sherwood McGowan
wrote:
>Remember guys, there's a LOT of IP blocks out there that are almost
>definitely not going to be somewhere you expect to receive SIP traffic
>from.
I agree. Is there a list I could use to check which blocks have been
allocated to which c
On 3/29/2011 12:25 PM, Steve Edwards wrote:
>> On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
>
>>> First thing I'd do is restrict the ip blocks your sip endpoints can
>>> register/call from in sip.conf (or your database's table for sip
>>> endpoints)
>
> On Tue, 29 Mar 2011, Gilles wrote:
>
Thanks again Tilghman
OK now I am back to the original.
[columns]
;static "" =>commented now
;alias => commented now
These are bogus and should never have been uncommented.
alias start => calldate uncommented now
alias callerid => clid uncommented now
Rest are all back to commen
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
First thing I'd do is restrict the ip blocks your sip endpoints can
register/call from in sip.conf (or your database's table for sip
endpoints)
On Tue, 29 Mar 2011, Gilles wrote:
Thanks for the idea, but it's not possible, as the Asteris
On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan
wrote:
>First thing I'd do is restrict the ip blocks your sip endpoints can
>register/call from in sip.conf (or your database's table for sip endpoints)
Thanks for the idea, but it's not possible, as the Asterisk must be
accessible for road war
On 3/29/2011 7:16 AM, Gilles wrote:
> On Mon, 28 Mar 2011 08:20:23 -0400, vip killa
> wrote:
>> Is anyone using asterisk with fail2ban?
> Sorry for hi-jacking the thread, but I was wondering if there were a
> lighter alternative that I could run on appliances?
>
> Python uses too much RAM, but I
On Tue, Mar 29, 2011 at 8:19 AM, Jonas Kellens wrote:
> Hello list,
>
> I'm trying this :
> *exten => _XXX.,n,Set(PY4=${CUT(PY2,\:,1-)}) *
> but this does not change a thing to the string...
>
Try the following:
*exten => _XXX.,n,Set(PY4=${CUT(PY2,:,2)}) *
--
Thanks,
--Warren Selby, dC
t38 from the itsp to asterisk using FFA = works
t38 from the spa8000 to the local asterisk using FFA = works (no
differance if FAX Passthru Method is set to reinvite or nse)
g711 to a remote system = works(i just turned off t38 on the
spa8000)
just the t38 pt is not working
th
Look at page 311 in that manual
If you disable the soft keys and then reassign the hard key it should
- at least in theory - be possible to accomplish.
On Tue, Mar 29, 2011 at 11:46 AM, C F wrote:
> Sorry, for some reason I misread it as the forward feature.
>
>
> On Sun, Mar 27, 2011 at 9:16 PM
Sorry, for some reason I misread it as the forward feature.
On Sun, Mar 27, 2011 at 9:16 PM, Mark Murawski
wrote:
> From the polycom pdf:
>
> divert.fwd.x.enabled
> If set to 1, the user will be able to enable universal call
> forwarding through the soft key menu.
>
> This sounds like it turns o
That did it. Thanks everyone!
On Tue, Mar 29, 2011 at 9:11 AM, Andrew Latham wrote:
> On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover wrote:
>> I have an Asterisk server running 1.6.2.13, where I can't seem to get
>> the increased logging to save to the /var/log/asterisk/messages file.
>> I have
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover wrote:
> I have an Asterisk server running 1.6.2.13, where I can't seem to get
> the increased logging to save to the /var/log/asterisk/messages file.
> I have tried using the standard "core set debug 10" and "core set
> verbose 10", as well as specifi
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dean Hoover
Sent: Tuesday, March 29, 2011 9:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Debugging not going to log f
I have an Asterisk server running 1.6.2.13, where I can't seem to get
the increased logging to save to the /var/log/asterisk/messages file.
I have tried using the standard "core set debug 10" and "core set
verbose 10", as well as specifically pointing it to the filename with
"core set debug 10 /var
On Tue, 29 Mar 2011 07:31:18 -0500 (CDT), Joe Greco
wrote:
>sshguard is *extremely* lightweight compared to most things; it's a very
>efficient compiled C application that doesn't have (m?)any dependencies.
Thanks much for the tip. I'll study how to install/configure iptable
and sshguard.
--
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Hello list,
I want to get the phone number out of the following P-Asserted-Identity
header :
/"BlaBlaBla" "/
I do the following in the dialplan :
/exten => _XXX.,n,Set(PY=${SIP_HEADER(P-Asserted-Identity)})
exten => _XXX.,n,Set(PY2=${CUT(PY,@,1)})/
This gives me :
/"BlaBlaBla" _XXX.,n,Set
On Tue, Mar 29, 2011 at 01:59:54PM +0200, Daniel Pocock wrote:
>
>
> >> - upgrade policy - is it intended that someone who has Debian 6 with
> >> the existing Asterisk 1.6 packages (from Debian's maintainer) can just
> >> upgrade to the Digium package without moving or changing any config?
> >
>
> On Mon, 28 Mar 2011 08:20:23 -0400, vip killa
> wrote:
> >Is anyone using asterisk with fail2ban?
>
> Sorry for hi-jacking the thread, but I was wondering if there were a
> lighter alternative that I could run on appliances?
>
> Python uses too much RAM, but I need to find a way to ban hackers
On Mon, 28 Mar 2011 08:20:23 -0400, vip killa
wrote:
>Is anyone using asterisk with fail2ban?
Sorry for hi-jacking the thread, but I was wondering if there were a
lighter alternative that I could run on appliances?
Python uses too much RAM, but I need to find a way to ban hackers from
trying to
>> - upgrade policy - is it intended that someone who has Debian 6 with
>> the existing Asterisk 1.6 packages (from Debian's maintainer) can just
>> upgrade to the Digium package without moving or changing any config?
>
> There is nothing specific about the packages that is going to make this
>
On Tue, Mar 29, 2011 at 12:23 PM, DHAVAL INDRODIYA
wrote:
> design your dial-plan for routing a specific number to different context ,
> you can try func_odbc for query to DB if you have a large number of setup.
> ideally its called click to call but you are made it as, miss call and you
> will ge
Hi All,
I am having some issues with Asterisk 1.8.3 extensions with a SIP Phone and
an gateway.
My setup is that I have my SIP Phone setup to register with the gateway.
Then the gateway should sent calls to the Asterisk as a type of friend.
This works fine if the SIP Phone configuration usernam
On Tue, 29 Mar 2011 07:48:08 +0200, wrote:
>I was a little unclear, it is not the cell phone that does the call-back, it
>is the cell-phone-network.
Makes more sense :-) Thank you.
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Dear All
I am using Asterisk 1.4.17 in a calling card application. Following
description explains the usage:
A call/request hits asterisk from an ip xxx.xxx.xxx.xxx which opens a
channel for this ip (Lets call it Channel A). Asterisk answers the call and
play IVRs first asking the PIN and then de
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