Hi list!
Today I tried to change the NAT-configuration on my Firewall to use
another port for SIP.
I configured it so:
/sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport 1:10100
-j DNAT --to-destination 192.168.20.120
/sbin/iptables -t nat -A PREROUTING -p udp -m udp --dport my
On Mon, Jun 8, 2015 at 9:56 AM, Igor Potjevlesch igor.potjevle...@gmail.com
wrote:
Hello!
I've got a little problem with Asterisk (11.14.1), the voicemessages are
kinda limited to 40 seconds (average) aproximately; because when a message
reach this long I got a cut in the file (*.wav) after
On Thu, Jun 4, 2015 at 5:58 AM, Tony Mountifield t...@softins.co.uk wrote:
Hi, despite some searching I haven't found an answer to this question:
Is there a way I can see in the log, or find out in the dialplan, what
codec has been negotiated for a SIP channel? If possible, I'd like to
do
Hi
Is there any way to set the presence state of a peer to in-use in asterisk
1.8?
The idea is to integrate DND buttons on phones to BLF.
Regards
--
Ishfaq Malik
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)161 660 2350
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w:
You can use a custom device state to do it.
[dnd]
;DND Toggle
exten = *363,1,Answer()
same =
n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})})
same = n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1)
;DND On
exten = *78,1,NoOP(Turning DND On)
same =
Hi John
I needed a dialplan solution so thank you very much for the pointer!
Regards
Ish
On 9 June 2015 at 17:27, John Kiniston johnkinis...@gmail.com wrote:
You can use a custom device state to do it.
[dnd]
;DND Toggle
exten = *363,1,Answer()
same =
Hi list!
I'm working hard to securing my Asterisk...
Now I deleted all possibility to access the node as anonymous and every
call through the proxy will be checked (just known peers are allowed to use
it).
Furthermore, I restricted the registration of my home phones to the Network I
reserved for
1 - My SIP server (Asterisk) will have some SIP clients registered in its SIP
registrar. Let's say 6 SIP clients. In my project I have to implement a way
of a SIP client making a call to a number and all others 5 SIP clients ring.
That is, the others 5 SIP clients must receive the SIP
On Mon, Jun 8, 2015 at 12:00 PM, Christian christia...@runbox.com wrote:
Hi,
Sorry if off topic, but is anyone here on this list using it?
I am currently searching for a good router for my home network wich
supports SIP.
Many thanks!
You'll probably get more answers in a generic VOIP/SIP
