Hi!
I have strange requirement: a incoming call should be duplicated to two
outgoing calls (to two voice recorders). On the incoming channel we only
receive RTP, on the two outgoing channel we only send RTP.
I thought of:
incoming call
- originate: make outgoing call to recorder 1 and
Hello,
using asterisk 1.8.12.2 and realtime architecture with mysql.
I get the following message on CLI when changing the value in the strategy
/[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param:
Changing to the linear strategy currently requires asterisk to be
restarted.//
What do you mean with voice recorders? Voice mail, if nobody answers, or do
want to monitor calls?
jg
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New to Asterisk? Join us for a live introductory
[Intentionally ignoring the Reply-to header in this reply. And yes, this
is on-list]
On Tue, Mar 25, 2014 at 03:15:22PM +, A J Stiles wrote:
On Tuesday 25 Mar 2014, Digium's Asterisk Development Team wrote:
We apparently have a spam bot subscribed to the list and replying
*directly* to
On Wednesday 26 Mar 2014, Tzafrir Cohen wrote:
[Intentionally ignoring the Reply-to header in this reply. And yes, this
is on-list]
What if I wanted to reply to one of your messages off-list?
My message would end up in asterisk_unwanted.
And it did -- which is why I have added a special
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207?
or variations but that same 972595 is often present.
Can someone break down that dial string with an explanation? The 011 look like
an overseas call (from Americas), while the 972595XX is unclear...
--
972 is Israel
See: http://en.wikipedia.org/wiki/List_of_country_calling_codes#Ordered_by_code
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis
Sent: Wednesday, March 26, 2014 11:05 AM
To:
On 03/26/2014 10:05 AM, Michelle Dupuis wrote:
I see a lot of attempts by hackers to call
00972595301123or 011972595115207 or variations but that same 972595 is
often present.
Can someone break down that dial string with an explanation? The 011
look like an overseas call (from Americas),
Hi
The 11 bit is them thinking there's some prefix which will cause your PBX
to become an open relay. The number (97259) is a Palestine Mobile number.
These's a lot of hacking attempts coming from Palestine and this type of
number probably has some revenue generation properties to it.
Regards
http://en.wikipedia.org/wiki/Telephone_numbers_in_Israel
Looks like it a mobile in Palestine - sure someone from Israel can
tell us more
2014-03-26 16:05 GMT+01:00 Michelle Dupuis mdup...@ocg.ca:
I see a lot of attempts by hackers to call 00972595301123 or 011972595115207
or variations
The number is not important i think.
You need to block those country's you never use to connect to your asterisk
system.
I bet this call is made from palestine/israel too.
Best regards.
Emiliano
Enviado desde mi BlackBerry de Personal (http://www.personal.com.ar/)
-Original
Hi
It's possible in Asterisk 1.8 enable verbose only in one context or
extension?
thanks
Att,
*Rafael dos Santos Saraiva*
http://br.linkedin.com/pub/rafael-saraiva/52/aab/230
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Hi Everyone.
I am getting this error WARNING[31977][C-0009]: chan_sip.c:10657
process_sdp: Can't provide secure audio requested in SDP offer
From the sdp can anyone suggest why secure audio cannot be provided
v=0
o=- 6611325078116277019 2 IN IP4 127.0.0.1
s=-
t=0 0
On 26 Mar 2014, at 15:05, Michelle Dupuis mdup...@ocg.ca wrote:
I see a lot of attempts by hackers to call 00972595301123 or
011972595115207 or variations but that same 972595 is often present.
Can someone break down that dial string with an explanation? The 011 look
like an overseas
Thanks Joshua.
Submitted issue ASTERISK-23539 with the information.
I verified my pjproject is up to date and included the latest git log commit I
have just in case.
Dan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
If this is to 972 area code then the next digits should be 0X or 0XX but they
are not. This differs from what I found documented for that area code - I
thought someone from the region might add to the discussion. Not sure if this
reflected a premium service etc. (But someone jumped in with
On 26 Mar 2014, at 16:20, Michelle Dupuis mdup...@ocg.ca wrote:
If this is to 972 area code then the next digits should be 0X or 0XX but they
are not. This differs from what I found documented for that area code - I
thought someone from the region might add to the discussion. Not sure if
On 3/26/2014 12:20 PM, Michelle Dupuis wrote:
If this is to 972 area code then the next digits should be 0X or 0XX but
they are not. This differs from what I found documented for that area
code - I thought someone from the region might add to the discussion.
Not sure if this reflected a
Hi all,
I have a user who is reporting dropped calls at his site. We don't have
any other users complaining of this.
So far, this is what we know:
1. The manager bought all new Polycom phones. (POE)
2. They replaced the network switch with a POE version.
3. It's not just one or two of the
I would suggest starting with a packet capture of the SIP messages that
will include both call legs (i.e. capture at the Asterisk box). This
should tell you who initiated the hangup - the carrier side, the phone
side, or Asterisk.
On Wed, Mar 26, 2014 at 11:46 AM, Mike Diehl
Hello,
When I get a SIP INVITE as follows:
INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
To: sip:02XX@IP:5060
Contact: sip:1053212@IP:5060
Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com
CSeq: 102 INVITE
Date: Wed,
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.9.1-rc2
DAHDI-Tools-v2.9.1-rc2
dahdi-linux-complete-2.9.1-rc2+2.9.1-rc2
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
On Wed, Mar 26, 2014 at 1:14 PM, Mickael MONSIEUR
mickael.monsi...@gmail.com wrote:
Hello,
When I get a SIP INVITE as follows:
INVITE sip:s@10.1.0.191:5060 SIP/2.0
Max-Forwards: 69
From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18
To: sip:02XX@IP:5060
Contact:
On Wed, 26 Mar 2014 16:20:58 +
Michelle Dupuis mdup...@ocg.ca wrote:
If this is to 972 area code then the next digits should be 0X or 0XX
but they are not.
You never dial the local trunk prefix when you're calling
internationally.
--
C. Chad Wallace, B.Sc.
The Lodging Company
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chad Wallace
Sent: Wednesday, March 26, 2014 6:31 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Numbers hackers call
If this is to 972
I remember having to turn off STP or set portfast on some switch ports to
some phones due to the boot sequence and timeouts of some phones a long
time ago.
Does anyone know which phones, if any still suffer from these problems?
I am setting up a lab and want to introduce this problem for the
I've noticed that the Asterisk (v11) security log captures attempts do dial
without first authenticating, and places the number dialed into the accountid
field.
I'm trying to distinguish between failed attempts to register and attempts to
dial without registering, but the security log treats
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