We do something similar to this by logging a Local channel (eg:
Local/1234@AgentContext) into the queue that passes each call through a few
lines of dialplan code before going to the SIP extension.
Jordan
From: asterisk-users-boun...@lists.digium.com
Hi,
I have written some very simple dialplan logic for our call centre agent system
so that when we log an agent into the queue they login as something like:
Local/4...@roamingagent/n
We have the occasional problem whereby Asterisk sees an agent as Invalid:
Local/4...@roamingagent/n with
I've used FFA briefly but successfully on Asterisk 1.6.2 x64.
Jordan
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: 05 March 2010 17:00
To: Asterisk-Users
Subject:
We use Asterisk 1.6.2 with the latest Polycom firmwares on the VVXs with no
problem.
They key is the new bootblock polycom released a little while back.
If you download the new BootBlock, BootROM and SIP Firmware from
http://www.polycom.eu/support/voice/business_media_phones/vvx1500.html it
I've never used it but...
http://www.snapanumber.com/
Looks ok feature-wise - plus there's a free version to take for a test
drive.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Soderblom
Sent: 18 December 2006 14:46
To:
Dean,
A small Linux box will make a very effective router (and firewall if
required) and give load balancing/failover capabilities. I've done it in
the past (many moons ago!)
A link from my bookmarks: http://lartc.org/ - can be a little scary
depending on your knowledge of ip routing and linux
Hi,
I've got a setup whereby calls come into the asterisk server (1.2.7.1)
over a IAX2 trunk and into a dialplan that launches a php AGI script:
[live-full]
exten = _X.,1,Set(TIMEOUT(absolute)=0)
exten = _X.,2,NoOp(${EXTEN})
exten = _X.,3,DEADAGI(live-full.php)
exten = _X.,4,Wait,2
exten =
Possibly a silly question, but do you have php installed
and configured in apache?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alok
MohapatraSent: 31 October 2006 15:45To:
asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk web
interface is not parsing
I've just installed Asterisk using TrixBox 1.1 (previously has 1.0
installed and working).
All my sip trunks and iax trunks connect and can receive calls (there
are no phones connected to Asterisk - it's just used for incoming
automated services), but the problem is that the line is silent.
The
