On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to load
balance incoming calls over IAX2 trunks. If any trunk goes down the calls
traffic will be shared with other available trunks. When it gets Up the
yeah -- searching how to perform this magic ...
On Fri, Dec 13, 2013 at 2:29 PM, Steven Howes steve-li...@geekinter.netwrote:
On 13 Dec 2013, at 07:48, Muhammad Usman replyus...@gmail.com wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want to
load balance incoming
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with other available trunks.
When it gets Up the script is
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with other available trunks.
When it gets Up the script is
On Fri, 13 Dec 2013, Don Kelly wrote:
What's the value of load balancing multiple IAX trunks between the same
system pair? What resources are being balanced?
1) It's been said (but I've never experienced it) that IAX performs poorly
at high channel levels. Something about all those packets
On Fri, 2013-12-13 at 06:20 -0600, Don Kelly wrote:
On Fri, 2013-12-13 at 12:48 +0500, Muhammad Usman wrote:
Hi - I have 2 Asterisk servers connected using 05 IAX2 trunks. I want
to load balance incoming calls over IAX2 trunks. If any trunk goes
down the calls traffic will be shared with
Michelle Dupuis wrote:
Some more details...I noticed that the call is bridged, and audio goes
one way. However, the dial command still times out after 35 seconds
(approx), and exists non-zero.
While the channels are up, I did an core show channel xxx and found
Blocking in:
ast_waitfor_nandfds
Is
I tried transfer=no, transfer=yer, and transfer=mediaonly (with a reload
inbetween)same result
I agree it sounds like something either end is using the wrong IP/port address
somewhere in the call (yet signalling works fine).
Anything else to suggest? I was hoping for an externalip type
Some more details...I noticed that the call is bridged, and audio goes one way.
However, the dial command still times out after 35 seconds (approx), and
exists non-zero.
While the channels are up, I did an core show channel xxx and found Blocking in:
ast_waitfor_nandfds
Is this a bug? Or
Hi,
I made good experienes with Siemens Gigaset C610 IP. This model is about
90 Euro. Configuration via web interface. But encryption (SIPS/SRTP) is
*not* possible with this phones.
-Thorsten-
Am 11.12.2013 11:30, schrieb Mario Giammarco:
Hello,
I need to setup this configuration:
-
If I need to use SIP, from where to get the suitable firmware for these Cisco
IP Phones 7942G?
Be careful, not all versions of SIP firmware work with asterisk. I do have 8-3-1
(cmterm-7941_7961-sip.8-3-1)here and it works just fine with my 7961. Downloaded
somewhere. Version 9.x is broken,
Friends let me define the scenario please;
Scenario:
2 asterisk servers (A B) are connected using 05 IAX2 trunks between them.
The machine A is running asterisk Openvpn server in TUN mode (5 instances
with difference IP addresses for clients). The machine B is running
asterisk with 05 OpenVPN
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