a fixed
location and if it does, you want to simply challenge it as usual to
prevent toll fraud?
I have found that its best to mitigate an attack at ingress before it
even makes it to critical infrastructure (media gateways,
application/media servers, etc.).
-metik
Bill W. wrote:
Hey Metik
a relevant example).
-metik
Bill W. wrote:
Hey Metik,
Thanks so much for your insights and your help. And yes, I was able to
append the X-AUTH-IP header with no problem. But that didn't solve the
issue. After some more research, it appears that the problem isn't with
auth-calls at all.
I
I noticed a typo in my post that may easily confuse someone...
user id=7105551212 cidr=127.0.0.0/8//
should be:
user id=7105551212 cidr=127.0.0.0/8
-metik
Metik wrote:
Bill,
I think you would add this to the user profile in the directory. The
brian.xml example (located in ${confdir
to its speed
and flexibility since it is not a B2BUA).
Based on Mathieu's response (and he is definitely someone that would
know), it looks like you should be able to easily append a X-AUTH-IP
header (via OpenSIPS) containing the IP address of the endpoint and call
it a day.
-metik
Bill W wrote
I use a similar method (transfer to XML dialplan based on the value of
${enum_route_1}) to determine if the SIP URI is native to a particular
FS instance or if it needs to be sent off-net and it works well.
-metik
Michael Collins wrote:
On Fri, Dec 18, 2009 at 2:21 AM, Robin Vleij vi...@fx
to your FS box, you may
want to use tcpdump or ngrep along with screen.
tshark (tty/cli vesion of Wireshark) and sipgrep are also extremely
useful. The later requires ngrep and a couple perl modules but I believe
it is included with FS in the contrib or scripts directory--I forget which).
-metik
boxes
to create an ad-hoc cluster. I actually have session border controllers
that have this feature and use it quite often.
-metik
Bill W wrote:
Hey Metik,
Thanks for the reply, and the pointers for doing it with xml_curl.
I'll guess have to do that in the short term, but in my opinion
that the value of Negotiated Dtmf-relay is rtp-nte.
-metik
Yehavi Bourvine wrote:
Hello Ognjen,
From the tests I've done it is not so... When I set the profile to
use INFO, and a phone calls and asks for RFC2833 (phone-events in the
SDP) the FreeSwich ignores it (does not have phone
some issues when multple dtmf relay types are left enabled on a voip
dial peer. Also, there are some (older) IOS versions that have issues
with DTMF duration which cause digits to be misinterpreted by the
far-end (PSTN/POTS) but not ignored altogether.
-metik
Yehavi Bourvine wrote:
Hello
significant investments in time and resources to implement it in the
sofia sip stack at the moment.
-metik
Tim Uckun wrote:
I have read some of the archived emails about HA, loadbalancing,
failover etc and I am still a bit confused about how I could set up
some sort of resiliency with freeswitch.
My
outgoing display-ie
(config-if)#isdn outgoing ie caller-number
(config-if)#isdn outgoing ie called-number
-metik
Yehavi Bourvine wrote:
Hello,
We have a Cisco running IOS 12.4T used as our SIP-PRI gateway. On
the PRI there is a Nortel with Q.Sig. After a lot of configuration
trials I've
it in the past and it does support it.
It is not by any means feature rich but should suffice given his needs.
The other alternative is to install Cygwin-based build of the ISC DHCP
server.
-metik
Karl J. Vesterling wrote:
You'd think so wouldn't you...
Even the DHCP Server with Snow Leopard
of gear (dslams, routers, softswitches, SIP endpoints, etc.) when a
dedicated server was not available.
-metik
Jeff Lenk wrote:
Hi
I run the SolarWinds TFTP server alongside FS on my small installation -
works nicely!
Jeff
Dave Stevenson wrote:
Hi,
I have just about got FreeSwitch
He should be able to just use Additional Option to add option 150 (and
the associated IP address to which the TFTP server is bound).
Brian West wrote:
Some Cisco phones need DHCP option 150.
/b
On Nov 19, 2009, at 10:46 AM, Dave Stevenson wrote:
Metik,
thanks a lot for the tip, I
Using the API, any caveats with transferring a call (via uuid_transfer)
that has been placed on hold (via uuid_hold) without using uuild_hold
off before doing so? Is it even possible?
-metik
___
FreeSWITCH-users mailing list
FreeSWITCH-users
Brian,
That explains what I have been seeing... The console would freeze or
the xml-rpc request would never receive a response (trunk rev 15463).
-metik
Brian West wrote:
uuid_hold will send a HOLD indication to the end you're talking to ...
it will NOT put the person your talking
free to email my off list and I
can help you. The cost difference should be minimal unless you move to second
line (a/k/a naked, unbundled, or dedicated) DSL.
-metik
- Original Message -
From: Lars Zeb
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, October 26, 2009 1:37
and adequate CPU (since the majority of them use software
based queuing and packet fragmentation). The only caveat is that the degree of
success can vary between firmware versions. Some of the consumer (gaming) or
small business (VPN) grade routers work well (Linksys, DLINK, etc.).
-metik
Why not simply overwrite the value of the variable used throughout the
script...
-- xml_curl.conf --
...
param name=gateway-url
value=http://localhost/index.php?xhostname=myhost; bindings=dialplan/
...
-- index.php --
?
$_REQUEST['hostname'] = $xhostname;
...
- Original Message
Please note that this would essentially be taking Chris' suggestion a little
further but the effort involved would be minimal.
- Original Message -
From: Metik
To: freeswitch-users@lists.freeswitch.org
Sent: Friday, October 23, 2009 5:37 PM
Subject: Re: [Freeswitch-users
already thought of
this and its a case of RTFM.
-metik
- Original Message -
From: Metik
To: Metik ; freeswitch-users@lists.freeswitch.org
Sent: Friday, October 23, 2009 5:48 PM
Subject: Re: [Freeswitch-users] Hostname
Please note that this would essentially be taking Chris
Both support it. In the Grandstream, I believe it is called Early Dial (vs.
SNOM's Overlap Dialing). It can be problematic if you have a device somewhere
in the middle that doesn't support 484s.
-metik
- Original Message -
From: Anthony Minessale
To: freeswitch-users
sort of interop issue with the carrier involved that is wrecking
havoc with your particular application?
-metik
- Original Message -
From: Tihomir Culjaga
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, October 13, 2009 3:24 PM
Subject: Re: [Freeswitch-users] SIP
bridge() appears to be ignoring the absolute_codec_string channel variable
defined in the User Directory even though info shows that it is present.
Other variables, such as effective_caller_id_number seem to behave
correctly which leads me to believe that this may be a very minor bug.
In order
Oddly enough, I initially though that was the problem and enabled it without
any success...
freeswi...@noesis.metik.com sofia status profile internal
API CALL [sofia(status profile internal)] output:
Math,
That was it--thank you very much!
-Metik
- Original Message -
From: Mathieu Rene
To: freeswitch-users@lists.freeswitch.org
Sent: Monday, May 18, 2009 3:25 PM
Subject: Re: [Freeswitch-users] User Directory and Per-user(Channel)variables
absolute_codec_string needs
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