Ya I'm passed that part now. I have dahdi properly loading the card, and
both links are green. Asterisk recognizes the channels, but still shows the
span as down.
On Sun, Jun 13, 2010 at 6:35 PM, C F wrote:
> Not sure what version you are running but I'm still running 1.2x in
> 1.2 you can't b
On Mon, 14 Jun 2010, Gopalakrishnan A.N wrote:
Is it possible can we do PSTN call hunting in Asterisk.
For example, 4 PSTN line are connected in FXO ports, customer will dial
one number, if the number is busy will it transfer to another line? or
it should only be done in telco side? Pls a
Hi,
Is it possible can we do PSTN call hunting in Asterisk.
For example, 4 PSTN line are connected in FXO ports, customer will dial one
number, if the number is busy will it transfer to another line? or it should
only be done in telco side? Pls advise.
--
Thank you with regards,
Gopalakrish
Whats about log?
--
Vardan Harutyunyan,
Senior System Administrator
Enterprise Incubator Foundation
123 Hovsep Emin Street,
Yerevan 0051, Republic of Armenia
Tel: + 374 10 219735
Fax: + 374 10 219777
E-mail: i...@eif.am
www.eif-it.com
Adil Zaaraoui wrote:
> Deal list,
> I have a problem with as
At 01:06 PM 6/13/2010, you wrote:
>We use a combo of aastra 9133i and 57i's. Don't the user id and the
>extension HAVE to be the same? I had thought the aastra's used the
>extension as the SIP id to register.
So in your extensions.conf you need lines like:
exten => 123,1,dial(SIP/123_thisisAfunny
On Sunday 13 June 2010 15:06:52 sean darcy wrote:
> As I mentioned, I'm not inclined to mess with the secrets, too much
> hassle for users. That's why I'm considering deny/permit.
Clearly, this intruder isn't costing you enough money yet. If you ignore the
problem for a month, does that cost you
Not sure what version you are running but I'm still running 1.2x in
1.2 you can't bring up PRI outside asterisk, since the PRI (I'm
assuming layer 2+) part loads with Asterisk.
On Sat, Jun 12, 2010 at 10:51 AM, Voip Asterisk wrote:
> Ya i'm not even to the asterisk part yet. I'm still trying to
what options allow asterisk to send a new invite with digest authenticaion
credentials after a challenge by far end upon original invite having no
credentials?
Joel O'Brien
--
_
-- Bandwidth and Colocation Provided by http://ww
Deal list,
I have a problem with asterisk call routing.
I configured my asterisk in a way that it forwards calls to a viop provider
using IAX2 protocol.
some cell phones numbers are routed and others are not!!! but when i try
directly zoiper with my provider it works perfectly.
Before, my aster
On Sun, Jun 13, 2010 at 4:06 PM, sean darcy wrote:
> On 06/13/2010 01:59 PM, Dave Platt wrote:
>>> If you leave your asterisk box open to the world with passwords like
>>> you deserve to be hacked..
>>
>> Well, without making a moral judgment, I will agree that you are *going*
>> to be hacked
Ya 99% sure that isn't it since they were just pulled working off an AS5300
On Sun, Jun 13, 2010 at 4:27 AM, Doug Lytle wrote:
> Voip Asterisk wrote:
> >
> > Status: Provisioned, Down, Active
> >
> > specifically the "Down" part.
> >
>
>
> In my experience that usually means the provider hasn't
On Sun, Jun 13, 2010 at 04:06:52PM -0400, sean darcy wrote:
> As I mentioned, I'm not inclined to mess with the secrets, too much
> hassle for users. That's why I'm considering deny/permit.
>
> Does that solve my problem?
If you don't have users who need remote access. The issue at hand is
brut
On 06/13/2010 01:59 PM, Dave Platt wrote:
>> If you leave your asterisk box open to the world with passwords like
>> you deserve to be hacked..
>
> Well, without making a moral judgment, I will agree that you are *going*
> to be hacked if you do this!
>
> The O.P. seems to have made two (fairl
On 06/13/2010 02:07 AM, dotnetdub wrote:
>
> The trouble with whitelisting, or using iptables to block 5060 (in fact
> * is behind a router - 5060 is port forwarded) is that traveling
> employees wouldn't be able to register with inbound extensions. We set
> up our travelers so they
On Sunday 13 June 2010 13:46:36 Tzafrir Cohen wrote:
> On Sun, Jun 13, 2010 at 10:59:43AM -0700, Dave Platt wrote:
> > The O.P. seems to have made two (fairly common) mistakes:
>
> [snip]
>
> > - Used the user's extension number as the SIP user ID... and
> >thus making it easy to figure out wh
I'm wondering if anyone knows a good, stable C AGI library (* v. 1.4 and 1.6
compatible).
I've taken a look at CAGI and QUIVR but their latest code releases date back to
2006.
I've also seen a more recent project (wildpbx) dated 2009:
http://github.com/comradeb14ck/wildpbx/tree/master/libraries/a
On Sun, Jun 13, 2010 at 10:59:43AM -0700, Dave Platt wrote:
> The O.P. seems to have made two (fairly common) mistakes:
[snip]
> - Used the user's extension number as the SIP user ID... and
>thus making it easy to figure out which user IDs on which a
>password attack could be carried ou
Hi All,
Been having problem using the AMI, i've got this PHP script:
$socket = fsockopen("1.2.3.4","5038", $errno, $errstr, $timeout);
fputs($socket, "Action: Login\r\n");
fputs($socket, "UserName: amiadmin\r\n");
fputs($socket, "Secret: amiadminpassword\r\n\r\n");
fputs($socket, "Action: Comman
> If you leave your asterisk box open to the world with passwords like
> you deserve to be hacked..
Well, without making a moral judgment, I will agree that you are *going*
to be hacked if you do this!
The O.P. seems to have made two (fairly common) mistakes:
- Used a "secret" so obvious t
Voip Asterisk wrote:
>
> Status: Provisioned, Down, Active
>
> specifically the "Down" part.
>
In my experience that usually means the provider hasn't brought up the
PRI. Granted, I've never used anything beyond a single PRI.
Doug
--
Ben Franklin quote:
"Those who would give up Essential L
Look!
> refuses MusicOnHold personalized
> -- Started music on hold, class 'personnalised'
Can you see it?! Two typos. ;->
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? J
Hi Michael,
Can you show us the output from:
"moh show classes" and "moh show files" Command
Or try it to set a new exten after setting the language with:
exten => 12345,n,Set(CHANNEL(musicclass)=personalised)
Daniel
Am 13.06.2010 um 12:35 schrieb Mickael Monsieur:
> Hello,
> The MeetMe applic
Hello,
The MeetMe application refuses MusicOnHold personalized and skip always in
the default!
Have you any idea how to fix this?
-- Executing [028883...@default:1] Set("SIP/109.10.214.1-0002",
"CHANNEL(language)=fr") in new stack
-- Executing [028883...@default:2] Answer("SIP/109.10.2
Ok making a little progress, but still can't get them completely up:
pri show span 1
Primary D-channel: 24
Status: Provisioned, Down, Active
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
Logical Channel Mapping: 0
T200 Timer: 1
On Sat, Jun 12, 2010 at 10:50:07PM +0200, Mickael Monsieur wrote:
> because... I use it! But I do not use MeetMe with!
>
> What is the importance of providing binary packets if the conference (MeetMe
> app) is impossible without compiling ??
What version (of Debian? Of Asterisk?)
Official Debian
25 matches
Mail list logo
