Displaying 20 results from an estimated 1000 matches similar to: "RTP sent to remote internal IP"
2015 Mar 14
0
RTP sent to internal IP
Hello List,
I need your advise please.
I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP
UA (not Asterisk), both are behind NAT. That remote peer is configured with
nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP
address which is declared in the Connection Information (c) in the SDP,
obviously reaching nowhere. I need RTP to be sent to the
2014 Jun 27
4
Attack on Sip server.
Hi All.
Someone is attacking on my SIP server.
There are lot of requests coming in and I am not able to stop it because I
am unable to detect the IP address.
I used wireshark to capture the packets.
Although I am using very strong password for my SIP users but still is
there any way to drop these packets and stop this attack.
I tried dropping packet after matching some string (most of the
2014 Sep 12
1
compiling Asterisk
I am trying to compile the certified-asterisk-11.6-cert5 code and when I try to start it and then go into the console I am getting the error message "asterisk dead but subsys locked". Can anyone help with why this is happening? I have never seen this before.
This is a fresh install on a new server CentOS 6.5.
Thanks,
Scott Haley
IS Voice Projects Team
Edward Jones Investments
Phone:
2014 Mar 10
0
Asterisk 1.8.15-cert5, 1.8.26.1, 11.6-cert2, 11.8.1, 12.1.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
and 12.1.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of these versions resolve
2014 Mar 10
0
Asterisk 1.8.15-cert5, 1.8.26.1, 11.6-cert2, 11.8.1, 12.1.1 Now Available (Security Release)
The Asterisk Development Team has announced security releases for Certified
Asterisk 1.8.15, 11.6, and Asterisk 1.8, 11, and 12. The available security
releases are released as versions 1.8.15-cert5, 11.6-cert2, 1.8.26.1, 11.8.1,
and 12.1.1.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of these versions resolve
2015 Mar 04
0
RTP suppress during calls - Asterisk 1.8.*
Im facing some problems with RTP during queue agent calls.
Randomly during the call the agent can't hear the other side. This happens
for two or three seconds and the the call continue without problems.
The weird thing is that the recording for this call is fine, so both sides
are recorded without interruption. I can hear both sides.
When this problem happen, all agents that is on call get
2013 Aug 28
0
AST-2013-005: Remote Crash when Invalid SDP is sent in SIP Request
Asterisk Project Security Advisory - AST-2013-005
Product Asterisk
Summary Remote Crash when Invalid SDP is sent in SIP Request
Nature of Advisory Remote Crash
Susceptibility Remote Unauthenticated Sessions
Severity Major
2013 Aug 28
0
AST-2013-005: Remote Crash when Invalid SDP is sent in SIP Request
Asterisk Project Security Advisory - AST-2013-005
Product Asterisk
Summary Remote Crash when Invalid SDP is sent in SIP Request
Nature of Advisory Remote Crash
Susceptibility Remote Unauthenticated Sessions
Severity Major
2010 Apr 20
1
Portech MV-374 does not register
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
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<body bgcolor="#ffffff" text="#000000">
<font size="-1"><font face="Helvetica, Arial, sans-serif">Hello list,<br>
<br>
has anyone experience with the Portech MV-374 GSM-gateway ?<br>
<br>
I'm
2005 May 16
1
A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel
I just registered ID 0004283 at http://bugs.digium.com for the problem
described in subject (found when using a Linksys PAP2-NA). I don't know
where the proper forum is to discuss, so I'm hoping anyone interested will
read the bug and let me know your thoughts, either at bugs.digium.com,
here, or by emailing me directly (or, please suggest another forum that is
more appropriate).
As
2017 Jul 12
2
Copying received and sent RTP packets due legal obligations
Hi,
I am facing a problem where for legal obligations (LI) I have
to copy/mirror/forward the RTP streams for some selected call
to an external address/port and I have not found a way to do
it with built-in functionality. Do I miss something?
The basic requirements are:
* Raw RTP (no transcoding, header and payload as is)
* Direction (did it arrive at asterisk or was it sent)
* End
2006 Dec 28
1
one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI> rtp debug
RTP Debugging Enabled
-- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered
2014 Mar 10
0
AST-2014-002: Denial of Service Through File Descriptor Exhaustion with chan_sip Session-Timers
Asterisk Project Security Advisory - AST-2014-002
Product Asterisk
Summary Denial of Service Through File Descriptor Exhaustion
with chan_sip Session-Timers
Nature of Advisory Denial of Service
Susceptibility Remote
2014 Mar 10
0
AST-2014-002: Denial of Service Through File Descriptor Exhaustion with chan_sip Session-Timers
Asterisk Project Security Advisory - AST-2014-002
Product Asterisk
Summary Denial of Service Through File Descriptor Exhaustion
with chan_sip Session-Timers
Nature of Advisory Denial of Service
Susceptibility Remote
2014 Mar 10
0
AST-2014-001: Stack Overflow in HTTP Processing of Cookie Headers.
Asterisk Project Security Advisory - AST-2014-001
Product Asterisk
Summary Stack Overflow in HTTP Processing of Cookie Headers.
Nature of Advisory Denial Of Service
Susceptibility Remote Unauthenticated Sessions
Severity Moderate
2014 Mar 10
0
AST-2014-001: Stack Overflow in HTTP Processing of Cookie Headers.
Asterisk Project Security Advisory - AST-2014-001
Product Asterisk
Summary Stack Overflow in HTTP Processing of Cookie Headers.
Nature of Advisory Denial Of Service
Susceptibility Remote Unauthenticated Sessions
Severity Moderate
2023 Jul 07
0
Asterisk Release certified-18.9-cert5
The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert5.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert5
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The following security advisories were resolved in this release:
2023 Jul 07
0
Asterisk Release certified-18.9-cert5
The Asterisk Development Team would like to announce security release
Certified Asterisk 18.9-cert5.
The release artifacts are available for immediate download at
https://github.com/asterisk/asterisk/releases/tag/certified-18.9-cert5
and
https://downloads.asterisk.org/pub/telephony/certified-asterisk
The following security advisories were resolved in this release:
2014 Sep 06
0
Certified Asterisk 11.6-cert5 Now Available
The Asterisk Development Team is pleased to announce the release of
Certified Asterisk 11.6-cert5. This release is available for immediate download
at http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-11.6-cert5
Thank you for your
2019 Nov 21
0
Asterisk 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5 Now Available (Security)
The Asterisk Development Team would like to announce security releases for
Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
These releases are available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk/releases