Displaying 20 results from an estimated 24 matches for "kharwel".
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kharwell
2020 Feb 25
2
pjsip startup errors when using "with-ssl" configure option
...o have the encryption support, what is it actually used for?
Maybe it is some old flag which is not needed anymore and so can be ignored
for now and possibly removed from the configure/makefile stuff for future
releases?
Kind regards,
Patrick Wakano
On Wed, 26 Feb 2020 at 06:33, Kevin Harwell <kharwell at digium.com> wrote:
> On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote:
>
>> Hello list,
>> Hope you are all doing well!
>>
>> I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box
>> and I wonder if someon...
2014 Jun 12
0
AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework
...Reported By John Bigelow <jbigelow AT digium DOT com>
Posted On June 12, 2014
Last Updated On June 12, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name CVE-2014-4045
Description A remotely exploitable crash vulnerability exists in the
PJSIP channel driver's pub/sub framework. If an attempt is
made...
2014 Nov 21
0
AST-2014-018: AMI permission escalation through DB dialplan function
...Reported By Gareth Palmer
Posted On 20 November, 2014
Last Updated On November 20, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name Pending
Description The DB dialplan function when executed from an external
protocol (for instance AMI), could result in a privilege
escalatio...
2014 Jun 12
0
AST-2014-005: Remote Crash in PJSIP Channel Driver's Publish/Subscribe Framework
...Reported By John Bigelow <jbigelow AT digium DOT com>
Posted On June 12, 2014
Last Updated On June 12, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name CVE-2014-4045
Description A remotely exploitable crash vulnerability exists in the
PJSIP channel driver's pub/sub framework. If an attempt is
made...
2014 Nov 21
0
AST-2014-018: AMI permission escalation through DB dialplan function
...Reported By Gareth Palmer
Posted On 20 November, 2014
Last Updated On November 20, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name Pending
Description The DB dialplan function when executed from an external
protocol (for instance AMI), could result in a privilege
escalatio...
2017 Apr 04
0
AST-2017-001: Buffer overflow in CDR's set user
...Reported On March 27, 2017
Reported By Alex Villacis Lasso
Posted On
Last Updated On April 4, 2017
Advisory Contact kharwell AT digium DOT com
CVE Name
Description No size checking is done when setting the user field on a
CDR. Thus, it is possible for someone to use an arbitrarily
large string and write past the end of the user fie...
2017 Nov 08
0
AST-2017-011: Memory leak in pjsip session resource
...Reported On October 15, 2017
Reported By Correy Farrell
Posted On
Last Updated On October 19, 2017
Advisory Contact kharwell AT digium DOT com
CVE Name
Description A memory leak occurs when an Asterisk pjsip session object
is created and that call gets rejected before the session
itself is fully established. When this happens the...
2018 Feb 21
0
AST-2018-002: Crash when given an invalid SDP media format description
...Reported By Sandro Gauci
Posted On February 21, 2018
Last Updated On February 19, 2018
Advisory Contact Kevin Harwell <kharwell AT diguim DOT com>
CVE Name
Description By crafting an SDP message with an invalid media format
description Asterisk crashes when using the pjsip channel
driver because pjproject's sdp parsing algorithm fails to...
2018 Feb 21
0
AST-2018-003: Crash with an invalid SDP fmtp attribute
...Reported By Sandro Gauci
Posted On February 21, 2018
Last Updated On February 19, 2018
Advisory Contact Kevin Harwell <kharwell AT diguim DOT com>
CVE Name
Description By crafting an SDP message body with an invalid fmtp
attribute Asterisk crashes when using the pjsip channel
driver because pjproject's fmtp retrieval function fails to...
2018 Jun 11
0
AST-2018-007: Infinite loop when reading iostreams
...Reported By Sean Bright
Posted On June 11, 2018
Last Updated On June 11, 2018
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name
Description When connected to Asterisk via TCP/TLS if the client
abruptly disconnects, or sends a specially crafted message
then Asterisk gets caught in an infinite loop while trying...
2019 Sep 05
0
AST-2019-004: Crash when negotiating for T.38 with a declined stream
...Reported By Alexei Gradinari
Posted On September 05, 2019
Last Updated On September 4, 2019
Advisory Contact kharwell AT sangoma DOT com
CVE Name CVE-2019-15297
Description When Asterisk sends a re-invite initiating T.38
faxing, and the endpoint responds with a declined...
2017 Dec 22
0
AST-2017-014: Crash in PJSIP resource when missing a contact header
...Reported On December 12, 2017
Reported By Ross Beer
Posted On
Last Updated On December 22, 2017
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name
Description A select set of SIP messages create a dialog in Asterisk.
Those SIP messages must contain a contact header. For those
messages, if the header was not present and using the PJSIP...
2014 Nov 21
0
AST-2014-017: <font size="3" style="font-size: 12pt">Permission escalation through ConfBridge actions/dialplan functions</font>
...Reported By Gareth Palmer
Posted On 20 November, 2014
Last Updated On November 20, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name Pending
Description The CONFBRIDGE dialplan function when executed from an
external protocol (for instance AMI), could result in a
privilege...
2014 Nov 21
0
AST-2014-017: <font size="3" style="font-size: 12pt">Permission escalation through ConfBridge actions/dialplan functions</font>
...Reported By Gareth Palmer
Posted On 20 November, 2014
Last Updated On November 20, 2014
Advisory Contact Kevin Harwell <kharwell AT digium DOT com>
CVE Name Pending
Description The CONFBRIDGE dialplan function when executed from an
external protocol (for instance AMI), could result in a
privilege...
2020 Nov 05
0
AST-2020-001: Remote crash in res_pjsip_session
...Reported By Sandro Gauci
Posted On November 5, 2020
Last Updated On November 4, 2020
Advisory Contact kharwell AT sangoma DOT com
CVE Name
Description Upon receiving a new SIP Invite, Asterisk did not
return the created dialog locked or referenced. This
caused a “gap” between the creation of the dia...
2015 Aug 10
2
asterisk queue - skills based routing (patch updated)
Dne 6.8.2015 v 21:00 Sylvain Boily napsal(a):
> Hello,
>
> Le 2015-08-06 09:24, Marek Cervenka a ?crit :
>> hi,
>>
>> there is updated skills based routing patch for asterisk queue
>> please test if you have time
>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-17366?jql=text%20~%20%22skills%22
>>
>>
>
> You can find the latest
2018 Dec 07
2
Question on WebRTC configuration
In the asterisk wiki instructions for Configuring Asterisk for WebRTC clients...
https://wiki.asterisk.org/wiki/display/AST/Configuring+Asterisk+for+WebRTC+Clients
"To communicate with websocket clients, Asterisk uses its built-in HTTP daemon. Configure /etc/asterisk/http.conf as follows:
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
2020 Feb 21
2
pjsip startup errors when using "with-ssl" configure option
Hello list,
Hope you are all doing well!
I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and
I wonder if someone can put some light on it.
Log history short, install_prereq fails to install the packages (not sure
how important they actually are....): speexdsp-devel, gmime-devel,
uriparser-devel, iksemel-devel, uw-imap-devel, hoard
Then, I am running the following commands
2015 Mar 06
0
res_pjsip endpoint config object's 'identify_by' option needs new value "uri".
On Fri, Mar 6, 2015 at 2:06 PM, Dmitriy Serov <serov.d.p at gmail.com> wrote:
> Hello.
>
> Asterisk 13.2.
> I transfer configs from chan_sip to res_pjsip.
> In chan_sip i have "match_auth_username=yes" and have nothing in pjsip.
>
> I have a lot of endpoints and registrations on same SIP server. And it's
> problem in pjsip now. Is not it?
>
> I
2020 Jan 22
0
[asterisk-app-dev] ARI Get Channel Variable
On Wed, Jan 22, 2020 at 5:32 PM Phil Mickelson <phil at cbasoftware.com> wrote:
> I'm trying to get the Call-ID from the SIP HEADER using getChannelVar.
> When I pass SIP_HEADER() and anything as the variable I get Unable to read
> provided function. If use Call-ID I get Provided variable was not found.
>
> This is a connected call. Is it not possible to get SIP HEADER