Hi
I'm using asterisk 1.8
Does anyone know how to use the queue reload command. The built in help
doesn't really help.
queue reload {parameters|membe Reload queues, members, queue rules, or
parameters
Regards
Ish
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)84
On 1/8/15 2:24 AM, Stefan Viljoen wrote:
Hi guys
Thanks for the pointers - I'll look into the possible compromise scenario
though I've got no idea how I'll counter it -if- I manage to detect it...!
I've disabled prelinking (thanks Tony!) and I'll see if that helps.
Interesting thing I've now
Hi
queue reload(queue name) or queue reload all
for example
queue reload reception
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik
Sent: Thursday, January 8, 2015 2:10 PM
To: Asterisk Users Mailing List - Non-Comm
That's what I would have guessed but it's not working:
[ish@??? ~]$ asterisk -rx 'queue show axon-all'
axon-all has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s
talktime), W:0, C:0, A:2, SL:0.0% within 20s
Members:
AXON200 (realtime) (Not in use) has taken no calls yet
It would appear that you have the Asterisk server on a public IP address,
your two endpoints are behind a NAT, and you have rewrite_contact enabled
in pjsip.conf.
In which case, what you are seeing is correct. In order to be able to send
a call to an extension where it is behind NAT, Asterisk mus
I am following the instructions in
https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I am
trying to make a call from extension Alice (6001) to extension for Bob
(6002). When I make the call, I can hear the ringing on Alice's phone
(caller), but Bob's phone (callee) doesn't ring
Thank you for your note, Scott.
I set rewrite_contact=yes for both contacts, and I also had to do
remove_existing=yes because I had to remove the existing contact
information (max_contacts = 1 was preventing new contact information)
using pjsip
qualify demo-alice etc., after which the right IP add
Hi list , someone on the list has seen this type of connection
attempts in asterisk, fail2ban does not stop
2015-01-08 14:59:47] SECURITY[21515] res_security_log.c:
SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100@
For anyone interested, Allison Smith's AMA (not sure she's still around):
http://www.reddit.com/r/IAmA/comments/2rrb7m/iama_professional_telephone_voice_ama/
--
Jeremy Kister
http://jeremy.kister.net./
--
_
-- Bandwidth and Co
Well, I thought it worked, but it actually doesn't--I am able to get the
caller pick up the phone, but for some reason, I cannot hear anything on
either side no matter who does the calling. Again, my two SIP phones are on
the local 192.168.1.0/24 network (do not go over the Internet) and the
Asteri
>I would also start by putting an audit rule on the binary. Something like
this:
>auditctl -w /usr/sbin/asterisk -p war -k asterisk-bin
>then you can get a report on who modified it and when by using:
>ausearch -f /usr/sbin/asterisk
>Its a start, but eventually you might need to monitor even key
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