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
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
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
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:
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
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
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 -
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
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
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
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