On Wed, Dec 19, 2012 at 12:44 PM, Andrew White
and...@computersforall.com.au wrote:
Hi Satish/list,
** **
Looks like I spoke to soon.
** **
I have the following in my dialplan:
** **
*Dial(${QUEUEEXTS},${RINGTIME},U(queueControl^direct^CONNECTED))*
** **
And after
2012/12/19 Scott Huang gyration.hu...@gmail.com
Hi
I've saw some similar case in the mail list, but seems no standard
answers, so I decide ask here again.
Is there anyone see the message below ? I use asterisk(1.8.11-cert 9)
in my openbts2.8, and when I made a phone call, the
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote:
Hi
Can someone else please check the following:
We have installed asterisk 1.8.18.0 onto our development and test
servers. They were previously on 1.8.7.0
When an inbound call executes a queue, I can see in the logs that the
hold
On Tue, 2012-12-18 at 10:01 -0600, Jonathan Rose wrote:
Harish Mandowara wrote:
I have Asterisk server 1.8.19 with jabber enabled.
On the other side i have openfire server with asterisk-im enabled.
I have a doubt, whether my sip client connected with asterisk can
send message to
Hi All,
I am googling from few days back for a conference utility
which fulfill my below scenario ,Please give your suggestion to fulfill my need.
I have a scenario where leader is giving a lecture and other
participants are on mute mode...
At the end of conference, when QA session begins, is
Scott Huang wrote:
Hi
I've saw some similar case in the mail list, but seems no standard
answers, so I decide ask here again.
Is there anyone see the message below ? I use asterisk(1.8.11-cert 9)
in my openbts2.8, and when I made a phone call, the Asterisk CLI
poppd the following
Is there a doc somewhere that explains when to use loop start vs. kewl start in
certain situations? I've seen in a lot of posts that most people just say 'use
kewl start', but nobody really explains why. It seems that if kewl start
should be used 100% of the time then why is there even an
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote:
Hi
Can someone else please check the following:
We have installed asterisk 1.8.18.0 onto our development and test
servers. They were previously on 1.8.7.0
When an inbound call executes a queue, I can see in the logs that
the
I’ve built a custom application for our call center and am having one
problem. Unfortunately certain things happen whilst the agent has
the customer on hold which I’d like to work around. But I can’t work
out how to catch the actual hold event so I can do something about
it. From the console
Hi gurus
Some of my agents are in use with no call involved.
Members:
Local/0971230031@internal/n (In use) has taken no calls yet
Local/0972105500@internal/n (In use) has taken no calls yet
No Callers
Is there a workarround so solve this?
Wham am i doing wrong?
If i change
Hi,
I have a PSTN Asterisk box that's connected to other dialplan PBXes through
IAX2.
Recently this box was upgraded to 1.4.44 with the latest DAHDI version.
I've noticed that if one of the dialplan PBXes calls Congestion(), the PRI
will return ISDN code 34 (as its supposed to do).
However, the
You probably already know this, but 1.4x is very old (released in 2006) and
is officially end-of-life.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions
You might get more help or better behavior by updating to a newer more
current version of Asterisk, such as 1.8 which will be
In 1.4.43 I would see things from core show channels like
DAHDI/18/x
for line 18
in Asterisk 11 its
DAHDI/i4/
How do I get the line number back?
Jerry
--
_
-- Bandwidth and Colocation Provided by
In 1.4.43 I would see things from core show channels like
DAHDI/18/x
for line 18
in Asterisk 11 its
DAHDI/i4/
How do I get the line number back?
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt file:
* The PRI channels in chan_dahdi can no longer change the
14 matches
Mail list logo
