Displaying 20 results from an estimated 20000 matches similar to: "Asterisk security between two servers"
2008 Feb 24
2
DUNDi with two servers
Hi,
I'm having difficulties with using DUNDi between two servers. If it were
three I think I could control looping by limiting TTL, but with two I'm not
sure how to prevent a loop causing bad things to happen. I've tried ttl=1
but things still blow up.
The DUNDi configurations are pretty simple and work just fine in both
directions as long as only one of them is using the switch
2008 Mar 05
2
Passing variables between two DUNDi/IAX2 peers
Hi.
I am trying to pass a variable from one Asterisk PBX
to another.
I'm using DUNDi with IAX2. Is there a way to do it?
I tried the following but it fails.
On peer1:
[dundi-outgoing]
switch => DUNDI/priv
exten => s,1,Set(CDR(userfield)=test)
exten => s,2,Set(DUNDIVAR=${ARG1}#TEST)
exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.)
exten => s,4,Goto(${DUNDIVAR},1)
On
2007 Feb 14
0
Requested contexts didn't get merged
Hello,
I have two asterisk servers and I would like to merge their dialplans. I
thought DUNDi would be a natural choice.
I created the following configuration on the first server:
iax.conf
Code:
[dundi]
type=user
dbsecret=dundi/secret
context=dundi-local
dundi.conf
[general]
ttl=4
autokill=yes
cachetime=30
entityid=00:06:5B:8E:B0:08
secretpath=dundi
bindaddr=XXX.XXX.XXX.XXX
port=4520
2008 Mar 18
0
AST-2008-003: Unauthenticated calls allowed from SIP channel driver
Asterisk Project Security Advisory - AST-2008-003
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Unauthenticated calls allowed from SIP channel |
| | driver
2008 Mar 18
0
AST-2008-003: Unauthenticated calls allowed from SIP channel driver
Asterisk Project Security Advisory - AST-2008-003
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Unauthenticated calls allowed from SIP channel |
| | driver
2011 Jul 25
0
Is there a protocol to be used to communicate between different Asterisk Boxes to distribute load and tasks
Really I am thinking in something and I donot know if it can be used or not:
Is it possible to use the Database (to be located in a server) so whenever the agent is login via any server, this will be logged in the database, so all the queues in all the boxes can check with this database to see if the agent is logged in and if it is available or not.
Is it possible to be like this?
About DUNDi,
2009 Feb 26
2
Problems with Outbound Calls
Hi everyone!
I'm quite a newbie at this Asterisk stuff so please bear with me.
We've recently decided to start training in Asterisk via AsteriskNow!
Asterisk version is 1.4.18.1 through AsteriskNow! 1.02
The box we have is paired with a Digium TE110P and we've managed to get
it to the point where incoming calls via a single DID (from NTT Japan)
can be received and answered
2010 Mar 22
0
DUNDi Confusion
Dear community,
Please help. I've been looking around the internet (and in this great forum)
for help with DUNDi setup between servers (I'm using Elastix) and while I
can get my servers to lookup extensions on each other very well, I have not
been able to successfully make calls between servers. For my test
environment, I have 3 servers setup for now, and these are the steps I've
2008 May 22
0
SIP configuration issues
Apologies if this is a repeat: I trawled through the archives and couldn't
find a reasonable answer, so I'm asking here. I have an Asterisk install
connecting from behind a NAT device (DSL modem) to a SIP proxy (in my case,
Broadvoice). I have an sjphone softphone on a Windows PC also behind the NAT
device that connects to the Asterisk install, and using this setup I've been
pretty
2007 Aug 08
1
Method for scripting options specified in make menuconfig
I've been digging around and I haven't found a way to do this, but I have a
feeling I'll feel like an idiot because it's something I'm over looking.
Normally if I need to specify an additional option (such as different
language sound files) or I'm building an Asterisk server with a lean
configuration and need to remove some modules I do so with 'make
menuconfig'.
2008 May 22
0
/home/putnopvut/asa/AST-2008-007/AST-2008-007: AST-2008-007 Cryptographic keys generated by OpenSSL on Debian-based systems compromised
Asterisk Project Security Advisory - AST-2008-007
+------------------------------------------------------------------------+
| Product | Asterisk |
|--------------------+---------------------------------------------------|
| Summary | Asterisk installations using cryptographic keys |
| | generated
2007 May 17
5
DUNDi configuration problem
Hi peeps,
I've been struggling with DUNDi for a few days now and I can't seem to
make call from Asterisk A to Asterisk B. If I do a "dundi show peers",
it finds the other peer but I can't seem to make any calls. Can
anybody help me out here.
Here's the situation:
Machine 1: Debian with Asterisk 1.4.4 --> 192.168.1.103
Machine 2: AsteriskNOW --> 192.168.1.69
The
2004 Dec 22
0
Macro(dundi-dundi-test, ${ENTEN}) to return +101 on lookup failure ?
I'm looking at finding a way for my Macro(dundi-dundi-test,${ENTEN})
when I dial out on the dundi-test network to return a +101 to my
[dundi-test-out] context, if the number being dialed on the dundi-test
network does not exist, then I will route the call out using my pstn
or voip connection i have. I have a feeling it will have to be the
switch => DUNDi/dundi-test that will have to return
2010 May 12
3
SIP trunk between two Asterisk servers
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something obvious.
I followed the simple example at http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so Asterisk server 1 (192.168.250.111) sip.conf contains:
[interboxsip]
type=peer
host=192.168.250.112
2006 Mar 16
2
Feedback from VON expo! Info on * HA andPolycomphone!!
Great Email. I'm going to respond to some of the points.
"Q: What are the plans for HA?
A: With a configuration using DNS-SRV and DUNDi, you can create a
pretty resiliant setup now."
That's BS. Last time I checked, Asterisk's support of SRV was to only grab the first SRV entry. Period. If it doesn't try any more SRV hosts after the first fails, just exactly how
2010 Mar 09
0
DUNDI Sip authentication failure
Hi all, I'm new in asterisk and I got to set up a dundi config for my work.
I have 2 PBX for the test, the two PBX are in the same local network
PBX A : 192.168.199.23
PBX B : 192.168.199.21
my config files : (on PBX B , the config files on PBX A looks like it)
/etc/asterisk/dundi.conf
[general]
bind=192.168.199.21
port=4520
cachetime=5
ttl=32
autokill=yes
entityid=00:30:18:4C:33:53
2007 Sep 11
2
Asterisk 1.4.11, res_features.so, SegFault
Hi All,
I have a really strange issue occuring where if I run "show dialplan" or
"dialplan show" or "dialplan show parkedcalls", then asterisk dumps core.
It only appears to happen with contexts that are created within
res_features. I am able to display all my other dialplans, but, every
time I try to just do a normal "dialplan show" asterisk core dumps
2006 Jun 14
2
DUNDi Users
I have three Asterisk boxes.
Each has the following in dundi.conf:
180net => dundi_local,0,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial
180q => dundi_q_pbx1,1,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial
180q => dundi_q_pbx2,2,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial
180q => dundi_q_pbx3,3,IAX,dundi:${SECRET}@${IPADDR}/${NUMBER},nopartial
My iax.conf on all three
2011 Apr 20
1
allowguest=yes, how?
Hello,
I want that people from other servers like ekiga.net can make calls to
my users. When I do an "allowguest=no" then people from other domains
cannot call me. So I think I need "allowguest=yes".
Maybe something like this?
-------------
<default>
include => users
<dialout>
include => users
exten=_0.,1,Dial(SIP/trunk/0${EXTEN:1},30,tT)
<users>
2014 Apr 16
1
DUNDi with SIP Mapping
>From the reading and testing I have done it doesn't look like SIP supports
a username and password in the Dial string. I currently have the following
mapping.
priv => dundi-extens,0,SIP,
dundi:pass at 1.1.1.1/${NUMBER},nounsolicited,nocomunsolicit,nopartial
On the sending side I see
NOTICE[31598] chan_sip.c: Conflicting extension values given. Using 'dundi'
and not