Hello list,
is it save to send manager events from a remote website (php) to an
Asterisk-server ? Is there some tutorial on how to implement a tight
safety policy ?
Jonas.
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Use simple PHP's telnet classes for AMI.
If you need special security - use Stunnel (SSL tunnel) and iptables
on asterisk side for IP forwarding.
This all is really straight-forward, I doubt you need a tutorial
here.. Both stunnel and PHP Telnet have tutorials on how to accomplish
this. You just
Jose P. Espinal wrote:
Hi,
Maybe you could do something in shellscript too:
e.g.
asterisk -rx help | grep -ia something
That would behave just as describe in the suggestion (but it's easier to
do :P)
You could place that in a tiny shellscript, that takes the 'something'
as an
Hi,
Please avoid HTML-only mail.
On Sat, Apr 24, 2010 at 12:30 PM, Jonas Kellens
jonas.kell...@telenet.be wrote:
Hello list,
is it save to send manager events from a remote website (php) to an
Asterisk-server ? Is there some tutorial on how to implement a tight safety
policy ?
What do you
On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote:
Hi Guys,
On 04-23-2010 21:40, Nathan Clemons wrote:
SIP is just the control protocol, and can be negotiated over TCP or UDP. The
actual payload is done over RTP, which is a UDP-based protocol.
thanks, for both of you for pointing this
Doesn't surprise me if Microsoft tries sending RTP over TCP. Maybe their
engineers didn't know basics of VoIP when they were programming their
communication server.
Zeeshan A Zakaria
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On 2010-04-24 12:04 PM, Michael Graves mgra...@mstvp.com wrote:
On
Hi!
Doesn't surprise me if Microsoft tries sending RTP over TCP. Maybe
their engineers didn't know basics of VoIP when they were programming
their communication server.
Not quite - doing SIP over TCP rather then UDP is the right thing to do
(tm). It's just that everyone started out with UDP
On Fri, Apr 23, 2010 at 3:21 PM, ad...@3a.hu wrote:
i have to put an * between two other SIP gateways and due to some
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
(ocs) and that's about it. In
RTP stands for Real-Time Transport Protocol. TCP is not designed to deal
with real-time data transfer as it takes time to acknowledge packets and
re-send them if missing. All audio video data transfer happens in real time,
and it doesn't make any sense to retransmit missing packets. Real time
Hi all,
Is anyone here using Asterisk on Archlinux?
If so, was it much to do in order for it to work?
Do you also use Dahdi?
many thanks,
Christian
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On Sat, 2010-04-24 at 10:56 -0500, Michael Graves wrote:
On Fri, 23 Apr 2010 23:11:06 +0200, ad...@3a.hu wrote:
Hi Guys,
On 04-23-2010 21:40, Nathan Clemons wrote:
SIP is just the control protocol, and can be negotiated over TCP or UDP.
The
actual payload is done over RTP, which is a
Adobe Air and Adobe FMS are good examples of VoIP working flawlessly over
TCP. We are actually developing a flash phone which needs only TCP to
transmit both signal and audio.
-Bruce
On Sat, Apr 24, 2010 at 2:01 PM, Zeeshan Zakaria zisha...@gmail.com wrote:
RTP stands for Real-Time Transport
Adobe has not been successful in VoIP. I have worked for two companies who
were trying to make flash phones, and it always came down to the issue of
RTP over TCP. This is the primary reason there are no successfully working
flash phones out there though some companies are trying to offer service
Hi
thwe PrivacyManger app states thast you can use a context to match against
for the input , but gives no real examples or explaination, does anyone have
a an example context for this
Thanks in advance
Robb
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On Sat, Apr 24, 2010 at 21:19, Christian christia...@runbox.com wrote:
Hi all,
Is anyone here using Asterisk on Archlinux?
Yes and no, I do use it on Archlinux for testing purpose but not as a
server.
Arch linux is not built to be a server distro, unlike Debian that have extra
steps for
On Fri, Apr 23, 2010 at 4:34 PM, russian qwerty
russian.qwe...@gmail.com wrote:
Hello.
As I see, there is a lot of threads about jitter buffer... Maybe anybody
knows something about my case? Any help will be appreciate.
So, the problem with voice quality was completely solved, BUT some
Hi,
I look after this but have been very busy for months. Maybe you canhelp me test?
Thanks,
Gavin.
On 23/04/2010, Sean Brady sbr...@gtfservices.com wrote:
Not sure if this is the right place to ask, but what do we need to do to
get this patch merged? How can I help? I'm no dev, but I use
Hey all
What VoIP networking monitoring, asterisk monitoring tools would you
recommend? I started working with an IT company that insists on using DSL
with a Sonicwall router. The problem is that the clients are having sound
problems. The owner is convinced that it's the Asterisk box. In the 4
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
2. Has anyone had luck running asterisk phone systems over DSL?
3, Has anyone used sonic wall routers for qos over dsl.
The company I am consulting for would like to install asterisk boxes over
dsl with sonicwall routers.
I think DSL is 1/2 duplex and in most cases way to slow on the way Up for VOIP.
I use a sonic wall on a T1 and it works great. It even has some
features for tweaking VOIP.
Any time I have tried VOIP on a 1/2 duplex connection all the way up
to 7down and 1.5 up I have call quality issues.
At
hello,
mike mosier schrieb:
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
you could try smokeping or iperf but real monitoring of the dsl quality
isnt easy.
2. Has anyone had luck running asterisk phone systems over DSL?
we dont run asterisk itself over dsl, but
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