On Mon, Mar 23, 2015 at 12:25 AM, Dmitry Melekhov d...@belkam.com wrote:
19.03.2015 09:31, Dmitry Melekhov пишет:
Hello!
As I see there is dsp_drop_silence switch in confbridge.
Could you tell me how asterisk detects silence?
Is it possible to change silence level,
so, let's say some not
Remember that that zyxel 16 port switch is only 8 poe ports. If your phones
are 802.3af or 802.3at, you could look at the ubiquiti line of switches.
On Mar 13, 2015 9:34 PM, Brian Franklin bfrank...@ntginc.net wrote:
If your phones support PoE,
I have had huge success with Zyxel:
hi,
i'm facing problem with multiple calls to one agent when Local channels
are used
wireshark shows multiple invites to the agent's phone
used versions
asterisk 1.8/asterisk 13
agents are logged dynamically. interface state based on hints
queue configuration
...
ringinuse=no
autofill = yes
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC
endpoints registered on asterisk 13 it is a feature that definitely asterisk 13
should support .
the problems that i faced with this is the following and i hope i could get
Harel Cohen wrote:
Hello List,
I need your advise please.
I am trying to establish a call from Asterisk 1.8.15-cert5 to one remote SIP
UA (not Asterisk), both are behind NAT. That remote peer is configured with
nat=yes in my sip.conf but yet RTP packets are being sent to its internal IP
address
Hello,
on previous versions of asterisk, extension h and H make us know who
ended a call (caller or callee). In the last * versions, seems that only
h extension is used, as stated here
http://www.voip-info.org/wiki/view/Asterisk+standard+extensions
In the last versions, how do we know which
On Mon, Mar 23, 2015 at 8:55 AM, Gosmac gosee...@gmail.com wrote:
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC
endpoints registered on asterisk 13 it is a feature that definitely asterisk
13 should support .
the
On Mon, 23 Mar 2015 10:11:54 +, Lukasz Sokol wrote:
No, ethernet switch works at lower / physical / MAC layer, NAT is
'above'
that;
so as long as everything is OK with your TCP/IP settings everywhere,
a switch is entirely transparent to TCP/IP (or generally, when it's
encapsulated into
Hi , I'm having some problems with functions enable auto answer in
some Grandstream GXP 1405 , I have enabled this feature in the snom
821 phone and work gr8 , in the gandstream not work, I enable the
function on the phone
Allow Auto Answer by Call-Info: yes
Dialplan:
exten =
On 22/03/15 03:03, thufir wrote:
On Fri, 13 Mar 2015 20:33:13 -0500, Brian Franklin wrote:
If your phones support PoE,
I have had huge success with Zyxel:
http://www.amazon.com/ZyXEL-ES1100-16P-16-Port-Ethernet-Unmanaged/dp/B00
so how does a client pc find the server if there's no NAT? by IP
address?? That makes no sense, to me, if the switch isn't assigning
addresses.
Switches have a MAC table that keeps track of which MAC addresses are on
which ports. That's how they decide where to route packets.
On Mon, 23 Mar 2015, thufir wrote:
so how does a client pc find the server if there's no NAT? by IP
address?? That makes no sense, to me, if the switch isn't assigning
addresses.
The 'endpoint' (pc, softphone, mobile, desk set, etc.) 'finds' the
server's IP address when:
) You configure
On 23/03/15 16:37, thufir wrote:
On Mon, 23 Mar 2015 10:11:54 +, Lukasz Sokol wrote:
No, ethernet switch works at lower / physical / MAC layer, NAT is
'above'
that;
so as long as everything is OK with your TCP/IP settings everywhere,
a switch is entirely transparent to TCP/IP (or
In the Asterisk log I see:
---
[Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29]
--- SIP read from UDP:198.38.7.34:5065 ---
SIP/2.0 200 OK
To: sip:16046289...@sip.babytel.ca;tag=sd3D4swKRc
From: sip:16046289...@sip.babytel.ca;tag=as07c833c5
Via: SIP/2.0/UDP
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