2011/12/27 virendra bhati virbh...@gmail.com
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server it's
remote server of server
You can also try special extension hangup and manage your scenario
On Sat, Dec 24, 2011 at 12:44 PM, Sammy Govind govoi...@gmail.com wrote:
Hi,
Please see the Dial application documents from CLI, i.e core show
application dial. There is an option which will let you continue in the
DIal-plan
Can you give an example how to set these oprion ...
On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini ldard...@gmail.com wrote:
2011/12/27 virendra bhati virbh...@gmail.com
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except
so you can try with options of dial application
g: Proceed with dialplan execution at the next priority in the current
extension if the destination channel hangs up.
G([[context^]exten^]priority): If the call is answered, transfer
the calling party to the specified priority and the
Hello,
I need to capture the DTMF digits dialled by user on current connected calls
and store them in variable.
scenario:
Manual Call Transfer:
User A dialed to B
B answered the call and want to transfer the call to user C manually. User B
dials *2 to get the ring tone again and then
Yes, this is one of my entries:
[trunk1]
context=fromoutside
type=friend
deny=0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
2011/12/27 virendra bhati virbh...@gmail.com
Can you give an example how to set these oprion ...
On Tue, Dec 27, 2011 at 1:43 PM, Leandro Dardini
On Tue, Dec 27, 2011 at 6:33 AM, virendra bhati virbh...@gmail.com wrote:
Hi Sammy,
I did the same and start calling. And it's working find but Now I want to
the server max capacity with this script then what is the correct process..?
There is a nice tutorial on how you can do this in the
Le 27/12/2011 16:04, Tim Nelson a écrit :
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server except iptables ?
[...]
Odd nobody
Hi,
as the Logs say clearly you need to create an extension in default context
named service
[default]
.
exten = service,1,NOOP(Incoming call from SIPp)
.
Regards,
Sammy
On Tue, Dec 27, 2011 at 5:48 PM, virendra bhati virbh...@gmail.com wrote:
Hi list,
I have installed SIPp into my
Hi Sammy,
I did the same and start calling. And it's working find but Now I want to
the server max capacity with this script then what is the correct process..?
On Tue, Dec 27, 2011 at 6:36 PM, Sammy Govind govoi...@gmail.com wrote:
Hi,
as the Logs say clearly you need to create an extension
With deny you'll deny all IP
with permit you'll permit only your IP.
Yes, it is mandatory to define both deny and permit.
Leandro
2011/12/27 virendra bhati virbh...@gmail.com
okay,
So it is mandatory to define both permit and deny ?
if I will update like
[trunk1]
context=fromoutside
On 12/27/11 08:16, Ryan Wagoner wrote:
On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote:
I'm getting various codec related warnings after upgrading to 1.8. Did
I miss something in the UPGRADE file? Does Asterisk no longer transcode
8-)?
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server except iptables ? I want to stop
on the basis of sip.conf account only. bcoz
I suspect nobody responded because this topic has been discussed over and over
again. Search the mailing list archives.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday,
Thank you Leandro,
Now i am able to register with fix IP.
On Tue, Dec 27, 2011 at 3:10 PM, Leandro Dardini ldard...@gmail.com wrote:
With deny you'll deny all IP
with permit you'll permit only your IP.
Yes, it is mandatory to define both deny and permit.
Leandro
2011/12/27 virendra
Hi list,
I have installed SIPp into my server. But not able to used it properly.
how to configure with my server ? how to see logs on webpage ?
how to start call testing
when i start SIPp then found verious hits on myserver.
*CLI:- *
[Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785
On Fri, Dec 23, 2011 at 10:40 AM, Eric Wieling ewiel...@nyigc.com wrote:
I'm getting various codec related warnings after upgrading to 1.8. Did I
miss something in the UPGRADE file? Does Asterisk no longer transcode 8-)?
WARNING[11123]: channel.c:4909 ast_write: Codec mismatch on channel
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati virbh...@gmail.com wrote:
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account only. bcoz I can't apply iptables rules on server
hangup extension works once the call is terminated but I want to know the
status of call immediately after connected, cancelled, or rejected and so on.
thanks,
Kamlesh
Date: Tue, 27 Dec 2011 16:59:35 +0530
From: dhaval.it01...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re:
Hello,
After investing some time, I could come to know the reason for not getting the
data value is that if I use system command with any of asterisk cli command as
given below, data value is returned blank.
$output=system(/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d
/ |
We are running 1.8.8.0.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan Wagoner
Sent: Tuesday, December 27, 2011 8:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
okay,
So it is mandatory to define both permit and deny ?
if I will update like
[trunk1]
context=fromoutside
type=friend
http://0.0.0.0/0.0.0.0
permit=34.2.10.24
qualify=yes
So will it be fine or not ? Or it will get rest information from sip.conf
general section ?
On Tue, Dec 27, 2011 at 2:21
Hello
I use fail2ban, and works fine,
Regards
On Tue, Dec 27, 2011 at 1:54 AM, virendra bhati virbh...@gmail.com wrote:
Hi list someone is trying to hack my server . Is there any way by whcih I
can stop hacking of my server except iptables ? I want to stop on the basis
of sip.conf account
On 12/26/2011 10:05 PM, sean darcy wrote:
I've now set up tcp to connect for some home-office connections.
Home is 10.0.0, office is 1.8.8.0.
The home sip device is home-going-to-office, office device:
office-coming-from-home - home ip is 10.10.11.180
-- Called SIP/home-going-to-office/166
2011/12/27, Eric Wieling ewiel...@nyigc.com:
We are running 1.8.8.0.
Then the issue you're having differs from the one I had (which
appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8
respectively).
--
_
--
- Original Message -
Le 27/12/2011 16:04, Tim Nelson a écrit :
- Original Message -
On Mon, Dec 26, 2011 at 11:54 PM, virendra bhati
virbh...@gmail.com
wrote:
Hi list someone is trying to hack my server . Is there any way by
whcih I can stop hacking of my server
Hi list,
I have a set of 300 or so WAV files I was combining and playing
using playback/background in 1.4.X. Now that I have moved on to the 10.0
set, I understand that I can replace my 8 Khz mono files with virtually
unlimited Khz mono files (still no stereo, but a quantum leap
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: 32433885116 sip:32433885...@ip.ip.ip.ip;tag=74706
Via: SIP/2.0/UDP
On 12/27/2011 01:43 PM, Jonas Kellens wrote:
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: 32433885116 sip:32433885...@ip.ip.ip.ip;tag=74706
Via:
On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:
On 12/27/2011 01:43 PM, Jonas Kellens wrote:
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: 32433885116
On 12/27/2011 01:51 PM, Jonas Kellens wrote:
On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:
On 12/27/2011 01:43 PM, Jonas Kellens wrote:
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID:
On 12/01/2011 03:48 PM, Jan Blom wrote:
Hello,
I’m trying to setup an Asterisk (version 1.8.8) to do SRTP termination
and then send the call on to other servers, unencrypted. All the basics
work fine.
I want the Asterisk to do as little as possible with the RTP packets and
no transcoding. We
Change requirecalltoken from auto to no. 1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday,
Asterisk 1.4 after, or beginning with, 1.4.26 DOES know about call
tokens, so this must be upset about something else
John Novack
Danny Nicholas wrote:
Change requirecalltoken from auto to no. 1.4 has no knowledge of this
parameter so turning it on in 1.8 creates an incompatibility (IMO).
On Ter, 2011-12-27 at 13:34 -0600, Danny Nicholas wrote:
Hi list,
I have a set of 300 or so WAV files I was combining and
playing using playback/background in 1.4.X. Now that I have moved on
to the 10.0 set, I understand that I can replace my 8 Khz mono files
with virtually
Hi team,
On event of no answer in CDR the starttime and endtime of call remains the same.
Is there any way how can actually track call originate time and call end time.
Thanks
Vinod dharashive.
Sent from BlackBerry® on Airtel
--
Was missing the option searchcontexts=yes in voicemail.conf
Mike
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On 12/27/11 18:23, Olivier wrote:
2011/12/27, Eric Wieling ewiel...@nyigc.com:
We are running 1.8.8.0.
Then the issue you're having differs from the one I had (which
appeared and disappeared instantly when I upgraded to 1.8.7 and 1.8.8
respectively).
Upgrading to 1.8.8 DID NOT HELP
I'm
No, it makes no difference, on the other end is asterisk 1.4.39
and 1.8.8 is still giving me:
Executing [4@internal:1] Dial(SIP/11-0003,
IAX2/home_server:@192.168.141.1/4,30,rw) in new stack
-- Called IAX2/home_server:@192.168.141.1/4
[Dec 27 20:00:16] WARNING[16398]:
Yes Eric,
I read the archive and found that all guys was saying another open sources
project for protection on server like fail2ban. But I want security at
configuration level only. As *Leandro* suggest permit and deny option of
Sip.conf and *Carlos* suggest the naming process. like that someone
may this helps,
In cdr.conf, set endbeforehexten=yes
Regards,
Zohair Raza
On Wed, Dec 28, 2011 at 4:46 AM, Vinod Dharashive vdharash...@gmail.comwrote:
Hi team,
On event of no answer in CDR the starttime and endtime of call remains the
same.
Is there any way how can actually track call
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