Displaying 20 results from an estimated 7000 matches similar to: "AST-2011-013: Possible remote enumeration of SIP endpoints with differing NAT settings"
2014 Jan 15
2
Asterisk ignoring nat settings
Hello,
I have an asterisk box with a peer configured with nat=force_rport,comedia,
but asterisk keeps sending the audio to the private IP address and ignoring
the client peer nat settings.
If I check the "sip show peer extension", I see both symmetric RTP and
Force Rport are set to yes, but asterisk seems ignoring them.
Force rport : Yes
Symmetric RTP: Yes
Asterisk is behind a
2014 Jan 07
1
Asterisk NAT friendly settings
I'm asking about this scenario:
Asterisk(public IP) <--> Internet <--> Router (public IP) <--> SIP
client (private IP and NAT)
What settings in sip.conf will give this the best fighting chance of
working?
We already have nat=force_rport,comedia
2015 Jun 07
3
Curious problem with NAT
Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb:
> What settings have you got for directmedia?
>
> Could you try
>
> nat=force_rport,comedia
> directmedia=no
Tried. Peer always unreachable, call not possible... :(
Other idea?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2015 Jun 07
2
Curious problem with NAT
Ashwin Surendran <Ashwin.Surendran at now-health.com> schrieb:
> Have you tried NAT=force_rport ?
OK, tried...
I can transmit from my phone (aka: I hear my voice on another phone), but I'm
not able to receive data (aka: I cannot hear what I say on the other phone).
Other suggestion?
Thanks
Luca Bertoncello
(lucabert at lucabert.de)
2014 Nov 03
1
issue with NAT
First I am new to PBX so i might be doing something fundamentally wrong...
That being said I got a FreePBX 32bit stable 6.12.65.
I am having some issue with the NAT and sound, both phones are ringing
but there is sound, I had some talk on IRC:
<[TK]D-Fender> Note for elfranne's situation, : nat=force_rport,comedia"
should have returned the public IP the call arrived on, but it
2013 Aug 18
1
Asterisk SIP Trunk between two Asterisk Servers
Hi,
Am making a simple SIP trunk between two Asterisk server,
Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port
extensions.conf
[man02-trunk]
exten => _1X.,1,Dial(SIP/usman02/${EXTEN})
exten
2017 Dec 14
4
SIP trunks going to the wrong context
Hi all,
I'm trying to resolve a weird issue with SIP routing.
I have a number of SIP trunks, from a selection of providers, all of
which are registered in sip.conf:
[general]
context=default
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0
tcpenable=yes
tcpbindaddr=0.0.0.0
transport=udp
bindport=15060
srvlookup=yes
allowsubscribe=yes
2014 Apr 16
2
FW: clients unable to auth
Hi Guys,
Just new to Asterisk and am completely stumped. I have created two accounts
as instructed. Please see below for the config of the user accounts.
[Peter]
type=friend
host=IP address
disallow=all
allow=ulaw
allow=alaw
callerid=Peter <6004>
secret=XXXXXXX
context=default
port=9060
nat=force_rport,comedia
deny=0.0.0.0
2015 May 28
3
Peer is UNREACHABLE
Darryl Moore <darryl at moores.ca> schrieb:
> Ahh. Seen that before! That suggests to me that you don't have your
> sip.conf records setup right.
>
> What's your sip.conf look like?
Well, here what I wrote in my sip.conf:
register => 00493511111111:MYSECRET at pbxluca/00493511111111
register => 00493512222222:MYSECRET at pbxfax/00493512222222
register =>
2019 Jan 15
2
(NAT) direct media to host on local net when registering from external address
On Tue, 2019-01-15 at 09:00 -0700, John Kiniston wrote:
> How is your endpoint currently configured in asterisk?
It's configured as a chan_sip peer.
> Have you tried
> rtp_symmetric to see if the endpoint sends audio to asterisk if
> asterisk
> can send audio back to the client?
That would require using chan_pjsip wouldn't it? Not that I am opposed
to trying that. I
2010 Aug 18
3
Playing with sipvicious ..
... using it as a tool and understanding what it does...
So one part of it's toolset identifys valid SIP accounts - and I was under
the impression that alwaysauthreject=yes was supposed to stop this...
However, it sends a request for a highly probably non-existent account,
then sends requests for probably existing accounts and I guess compares
the results - account not found vs. bad
2017 Jan 24
2
Asterisk 13.13.1
Hello, I recently upgraded from Asterisk 1.8 to Asterisk 13. Now users are
starting to complaint about packets loss, conversations are choppy!
I don't even know where to start looking! Choppy conversations happened
within users. I am using sip.conf
[1091]
type=friend
context=sip-phone
call-limit=2
trustrpid=no
callerid="dev1" <1091>
disallow=all
allow=ulaw
2017 Dec 01
0
AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
Asterisk Project Security Advisory - AST-2017-013
Product Asterisk
Summary DOS Vulnerability in Asterisk chan_skinny
Nature of Advisory Denial of Service
Susceptibility Remote Unauthenticated Sessions
Severity
2014 Jan 21
3
Asterisk Fax detection *11.7
Hello everybody
I'm trying to enable the Digium res_fax app at my *11.7 Server.
a fax show stats comes up with
FAX Statistics:
---------------
Current Sessions : 0
Reserved Sessions : 0
Transmit Attempts : 0
Receive Attempts : 1
Completed FAXes : 1
Failed FAXes : 1
Digium G.711
Licensed Channels : 1
Max Concurrent : 0
Success : 0
Switched to
2015 May 28
4
Peer is UNREACHABLE
Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb:
> The phone you gave your wife is really old. Are you sure it supports SIP
> OPTIONS? Can you make a call in or out to it? If you can, it is more
> likely that it just doesn't support that and you can't use a qualify
> statement.
No, I'm not sure.
And no, I can't make any call, right now... At least,
2014 Nov 21
0
AST-2014-013: PJSIP ACLs are not loaded on startup
Asterisk Project Security Advisory - AST-2014-013
Product Asterisk
Summary PJSIP ACLs are not loaded on startup
Nature of Advisory Unauthorized Access
Susceptibility Remote unauthenticated sessions
Severity Moderate
2014 Nov 21
0
AST-2014-013: PJSIP ACLs are not loaded on startup
Asterisk Project Security Advisory - AST-2014-013
Product Asterisk
Summary PJSIP ACLs are not loaded on startup
Nature of Advisory Unauthorized Access
Susceptibility Remote unauthenticated sessions
Severity Moderate
2010 Jun 24
2
Friday at 1PM: SIPVicious has a new tool: svcrash
Hi,
Got some great news a few days ago from Sandro Gauci (@SandroGauci)
and we'll be talking about this with him this Friday at 1PM.
SIPVicious, the free security tools for SIP scanning, now include a
new tool: svcrash. It is aimed at helping system administrators stop
bandwidth consuming scans making
use of svwar and svcrack. Here is the announcement on SIPViscious blog:
2012 Aug 30
0
AST-2012-013: ACL rules ignored when placing outbound calls by certain IAX2 users
Asterisk Project Security Advisory - AST-2012-013
Product Asterisk
Summary ACL rules ignored when placing outbound calls by
certain IAX2 users
Nature of Advisory Unauthorized use of system
Susceptibility Remote
2010 Aug 30
1
Fail2ban integration issues with Asterisk 1.4.21 under Debian Lenny
Hi,
I've recently had a fairly prolonged SIP registration attack, 18 hours in
this case and often with 200 attempts per second, and suspect I've had a
number of these in the past. The main symptom I noticed previously was,
because Asterisk was responding to each registration request it received,
it was very quickly using up my 448 kbps upload limit for my home ADSL
connection: any