Hi Amit,
My rtp.conf has the stunaddr listed and icesupport set to yes.
It looks like the issue is that the media isn't being sent from 192.168.3.150
to 192.168.3.131 (chrome browser to asteriskrtc.local).
When using asteriskrtc.local to originate the call (make a call directly from
sipml
Dear All,
I have make a queue in my dailplan and queue is not working properly,prbolem is
that all call goes to same extenstion at a time.Because,I use
eyeBeam(softphone) and eyeBeam have six line and whenever a call comes into
eyeBeam that call reserved by Line 1 suppose to 2nd call will come
On 22 May 2014 12:42, omakhileshchand omakhileshch...@gmail.com wrote:
Dear All,
I have make a queue in my dailplan and queue is not working
properly,prbolem is that all call goes to same extenstion at a
time.Because,I use eyeBeam(softphone) and eyeBeam have six line and
whenever a call
I would research the ringinuse option as well.
On 22 May 2014 13:42, omakhileshchand omakhileshch...@gmail.com wrote:
Dear All,
I have make a queue in my dailplan and queue is not working
properly,prbolem is that all call goes to same extenstion at a
time.Because,I use eyeBeam(softphone)
In the past little while, we've seen
a wave of attacks on asterisk, via the
provisioning.
It goes something like this:
A. scan for IP phones on the internet,
either via spotting something on port 5060,
or via the port 80 web interface for the phone.
Or, use web sites that scan the
On 5/22/2014 12:41 PM, Steve Murphy wrote:
So, these defenses can be employed to stop/ameliorate such
hacking efforts:
1. Keep your phones behind a firewall. Travellers, beware!
Never leave the default login info of the phone at default!
2. Never use the default provisioning URL for the
For a sip-only application, what exactly is required to ensure that
calls completed via followme are reinvited? Can it at all?
The code after outbound = findmeexec(targs, chan) calls ast_bridge_
call(). I don't see anything there which can cause a reinvite, yes?
When the same peer is used for
On Wed, 21 May 2014 23:09:28 +0200
Bart Remmerie remme...@gmail.com wrote:
configure: *** The IMAP_TK installation appears to be missing or
broken.
[...]
These are the steps I followed:
sudo apt-get install libssl-dev libpam0g-dev
cd ~/src/asterisk-complete
mkdir third party
cd third
Hello,
We have servers running Asterisk 1.8.20.1 and 11.7.0, and on both setting
maxsecs in voicemail.conf doesn't seem to have any effect. A voicemail
keeps recording after the specified time, and when the caller hangs up the
voicemail is saved in the mailbox.
Are we doing something really
