Hello!
First of all, you should disable unused VoIP protocols. Than remove all
guest accounts from used protocols, disable guest unauth access.
Always use strong passwords for accounts, for users on your system.
Passwords shouldn't be eq username. Move port binds on LAN network for
all active
:
My asterisk box was hacked!
On Thu, 21 Jul 2011, Захаров Антон wrote:
First of all, you should disable unused VoIP protocols.
Once a box has been hacked you cannot trust anything.
Disconnect the box from the network, save whatever DATA ONLY you
cannot live without, DBAN the disk and start
I don't think so. I think it's a predefined pairs of TON and NPI, that
could not be set separately.
On 24.05.2011 04:35, Rafael dos Santos Saraiva wrote:
did not work!! Bug in Asterisk?? :(
Rafael
2011/5/20 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru
Yeap, I couldn't set Private
Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0)
NPI: Unknown Number Plan (0) '81747956' ]
I set Private TON, but display National TON.
Thank's
Att,
Rafael Saraiva
2011/5/19 Захаров Антон ins...@mail.ru mailto:ins...@mail.ru
Hello.
To apply this settings you should
Hello.
To apply this settings you should restart dahdi (dahdi restart in
CLI). About influence you could read here:
http://markmail.org/message/rpd2aewiu2soostz
On 19.05.2011 06:05, Rafael dos Santos Saraiva wrote:
Hi
I'm beginner in list. I have doubts about the options pridialplan and
Hello, all.
Could you tell me how to set the type of number in the outgoing SETUP
message sent over PRI trunk?
I need to have:
Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ]
But always have:
Called
Try to look in 'sip set debug peer user'.
On 29.04.2011 18:10, Mike wrote:
Hi,
I have been getting reports phones ringing only a tiny moment and then
going to voicemail. CLI output shows:
-- SIP/user-0006fcdd is ringing
-- Got SIP response 400 Bad Request back from 23.23.23.23
I have enabled DNS manager in /etc/asterisk/dnsmgr.conf. It helps me.
On 19.04.2011 14:05, Niccolò Belli wrote:
Il 18/04/2011 12:22, Alexandru Oniciuc ha scritto:
Disable DNS lookups. Chan_sip crashes asterisk if you have that enabled and
internet is offline.
srvlookup = no didn't help.
Could 'dnsmgr' help?
On 18.04.2011 14:30, A J Stiles wrote:
On Monday 18 Apr 2011, Niccolò Belli wrote:
As soon as the Internet connection goes down, phones
stop working. I want to be able to use pstn, isdn and the gsm gateway
even if the Internet connection goes down, how can I achieve it?
Hello!
Try to use ${CHANNEL} instead of agi_type.
It will be like this:
$typ = $AGI-get_variable('CHANNEL');
@tmp_array=split(/\//, $typ);
$typ = $tmp_array[0];
and
$src=$AGI-get_variable('cdr(src)');
On 05.03.2011 10:25, Olivier CALVANO wrote:
Hi
i want use the API on my asterisk 1.6,
On 11.02.2011 12:37, Ishfaq Malik wrote:
Hi
Does anyone have any rough idea how far away 1.8.3 is?
We can't deploy 1.8 yet because of this issue
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=18403
Have you tried issue18403.patch ?
--
On 26.11.2010 17:29, David Backeberg wrote:
2010/11/25 Захаров Антонins...@mail.ru:
Hello everyone.
I have a timing slips errors and I can't understand what source of the
problem is.
My installation has 2 digium cards: TE420 and TE220 cards in one server.
There are 3 spans (E1) to PSTN and 3
нас он соединяет две 4-портовые карточки. Не помню, что именно было до
того, как его поставили, но какие-то проблемки были.
25.11.2010 19:23, Захаров Антон пишет:
Hello everyone.
I have a timing slips errors and I can't understand what source of the
problem is.
My installation has 2 digium cards
Hello everyone.
I have a timing slips errors and I can't understand what source of the
problem is.
My installation has 2 digium cards: TE420 and TE220 cards in one server.
There are 3 spans (E1) to PSTN and 3 spans to internal PBS stations -
normal installation for transit communication.
Span
[ivr_holiday]
switch = Realtime/ivr_holid...@extensions
where 'ivr_holidays' is context and 'extensions' is table
On 01.10.2010 12:52, Phibee Network Operation Center wrote:
Hi
i am not a expert on Asterisk and search a lot of small information :
I use Asterisk 1.6.1.4 with
Hello everyone.
I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri
1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from
telco. Another trunk looks to PBX with DECT system.
Some outgoing calls from asterisk to PSTN drops. The last message that
exists before
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