Greetings!
I've been playing around with "clustering" some
Asterisk servers for sake of fail-over and load balancing with DNS
round-robin, and came to one problem.
If I have, say, 2 servers, and
clients register either on 1 or 2, how can I route extensions between
them? I mean, if today user
The way I accomplish this is by having an active/passive cluster. The two or
more servers have individual IP addresses and running heartbeat creates a
clustered IP address. The active server uses the cluster IP address. If the
active server should fail then the cluster IP address moves to ano
Hello,
We use AMI to originate calls. Sometimes, lately every morning, the AMI
Originate process operates extremely slowly. I cannot see the calls in "core
show channels verbose", I don't know where they are, what state they are in,
after 2-3 minutes the calls go through one after the other. A
Hi Dave,
On Fri, May 18, 2012 at 11:27 PM, Dave Platt wrote:
>
>> In our app we do not forward packet immediately. After enough packet
>> received to increase rtp packetization time (ptime) the we forward the
>> message over raw socket and set dscp to be 10 so that this time
>> packets can escape
On Saturday 19 May 2012, Mikhail Lischuk wrote:
> I've been playing around with "clustering" some
> Asterisk servers for sake of fail-over and load balancing with DNS
> round-robin, and came to one problem.
>
> If I have, say, 2 servers, and
> clients register either on 1 or 2, how can I route ext
I'm in the process of setting up an asterisk box that will stand
between PBX's and the SIP providers. So a trunking server.
How can I 'test' to see if this trunking server is stepping out of the
media path during calls?
Thanks
David
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www.ringfree.biz
828-575-0030
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tcpdump and wireshark would help I guess. Just sniff for sip traffic and look
out for what's happening there. My 2 cents
Sent from my iPhone
On May 19, 2012, at 8:33 PM, David Wessell wrote:
> I'm in the process of setting up an asterisk box that will stand
> between PBX's and the SIP provider
Olivier wrote:
> Hi,
>
> At the moment, I'm mostly using a "Day/Night toggle" button to let
> users deal with week-ends, holidays and opening hours.
> As Asterisk 1.8 introduces Calendar capabilities, I'm wondering if
> better alternatives now exist.
>
> Is it possible, safe, reliable and easy to r
is it possible to use data voice dial-up modem to make and receive call for ivr
system..
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New to Asterisk? Join u
I use an SBC to protect my pool of asterisk servers and as trunking
endpoint with SIP Telcos. Now I'm trying to implement SIP phone
registration to be delegated through the SBC, as a proxy.
It doesn't work. It just works when I don't use realtime peers at the
asterisk servers. Using realtime SIP p
Mahendra Dobariya wrote:
is it possible to use data voice dial-up modem to make and receive
call for ivr system..
This should help:
http://www.voip-info.org/wiki/view/X100P+clone
Doug
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Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
d
Hey Mahendra,
We happen to still stock the same X100P cards in India.
If you require one, do connect with me off the list.
Regards,
Mitul Limbani,
Chief Architech & Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.ente
Greetings List.
I Have a small test server and i'm facing a small issue.
i have setup two SIP PEERS and they are able to do Video calls.
now I'm testing SET SIP_CODEC in a dial plan and when ever i'm setting the
codec .. the inbound (=first) leg stops receiving or sending video and SIP SHOW
CH
Hi all,
I have two asterisk servers A and B.
And I would like from A, dial to B passing some IAX variables.
Then B handles the calls, setup some other variables that become available
to A which can continue.
So far, I have used IAXVAR function.
It works when sending call from A to B
But varia
Uhm, if the dialplan is exactly as you pasted, you're not setting TESTVAR2 to
anything. You would need some sort of Set(IAXVAR(TESTVAR2)=...)
Noah
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ruddy Gbaguidi
Sent: Saturday, May 19,
Sorry, the dialplan is really on server B
exten => s,n,Set(IAXVAR(TESTVAR2)=efgh)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah
Engelberth
Sent: 2012-05-19 14:45
To: Asterisk Users Mailing List - Non-Commercial Discussi
Thanks Vladimir for the response and apologies for my extremely late
one! This week was quite crazy!!
I have just upgraded my server to FreePBX to take advantage of an
"up-to-date" system.
I migrated the original config over with a few set changes.
On 05/14/2012 04:42 AM, Vladimir Mikhelson
Of course you are disabling the video maybe also include the video protocols in
the sip_codec
-Original Message-
From: Tarek Sawah
Sender: asterisk-users-boun...@lists.digium.com
Date: Sat, 19 May 2012 17:33:57
To: Asterisk Users
Reply-To: Asterisk Users Mailing List - Non-Commercial D
Thank you, Any idea how?
Need to be able to control the codecs in use
through soem bandwidth tests. so i need to be able to set the SIP_CODEC
and still be able to do Video.
any suggestions?
> To: asterisk-users@lists.digium.com
> From: isr...@gmail.com
> Date: Sat, 19 May 2012 20:38:22 +
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