On 4/25/2011 9:38 AM, David Vossel wrote:
If you are already familiar with ConfBridge from Asterisk 1.6.X and
1.8, forget everything you know. This is a completely revamped,
highly optimized, and feature rich conferencing application capable
Can you give a quick lesson on how to use
Thank you all for your answers,
I will stick with nat=yes.
Alex
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Hi,
wrandom strategy for Queue says...rings random interface, but uses the
member's penalty as a weight when calculating their metric. So a member
with penalty 0 will have a metric somewhere between 0 and 1000, and a member
with penalty 1 will have a metric between 0 and 2000, and a member with
Hi AM,
I tried this on Asterisk 1.8.0 and found different behaviors each time.
Isn't that part of the definition of random? If Asterisk would
behave the same each time it wouldn't be random but predictable, I
would say.
AFAIK the metric just means that you get a higher or lower chance of
being
Thanks Jaron,
I understood the point from your explanation.
What should I do if I always want to ring a particular Queue member first
whenever he is available?
Yes, I can dial that member first before sending the call to Queue and
achieve the result but just wanted to know views from others.
Dear all,
I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6.
When I am executing following AMI originate API. Orginate start to
execute extenstion without knowing of PSTN(FXO) channel is ringing.
Any one can help me to resolve this issue ?
Action: Originate
Channel:
Originate successfully queued only means that the originate action was handed
off to asterisk not that is was executed yet. There are other events, depending
on which events you are reading, that tell you the call was answered and such.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
Hello.
Considering the following setup:
Legacy PBX --(ISDN)-- Asterisk --(MFC/R2)-- PSTN
When a user dials out, Asterisk receive overlap digits, matches them to an
extension and dial the PSTN, completing the call. So far so good.
The issue I'm trying to solve (or at least improve) is the
Hi All;
I am using Asterisk 1.8, how I can protect my self from hackers in case they
was able to see my sip.conf file? I need the password to be encrypted, how?
Regards
Bilal
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-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of bilal ghayyad
Sent: Tuesday, April 26, 2011 9:14 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Password to be ecrypted?
Hi All;
I
On Tuesday 26 Apr 2011, bilal ghayyad wrote:
Hi All;
I am using Asterisk 1.8, how I can protect my self from hackers in case
they was able to see my sip.conf file? I need the password to be encrypted,
how?
Short answer: You can't. Asterisk itself needs to be able to read the stored
Hello List,
Please help with the following problem,
I have a situation, where I need to play an audio announcement to the caller
SIP channel once an attended transfer is successful. The attended transfer is
done from client. I can see a transfer event in AMI. I am not using 'T/t'
option in
Hi All,
I have an strange behaviour, sometime (so far I am not sure how to reproduce
the problem) when I call to a meetme room, the system asks me for the pin and
after that what I can hear is a sound like an ambulance siren.
After restarting the asterisk process everthing works again.
The
The Asterisk Development Team has announced the release of Asterisk 1.4.40.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
After the initial release of AST-2011-006, a regression was found and then
resolved. This release contains the
The Asterisk Development Team has announced the release of Asterisk 1.4.41. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.4.41 resolves several issues reported by the community
and would have not been possible
The Asterisk Development Team has announced the release of Asterisk 1.6.2.18.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of Asterisk 1.6.2.18 resolves several issues reported by the
community and would have not been
Which version of 1.4 is current, 1.4.41 or 1.4.40.2? I received both just now
and actually received the notification for 1.4.40.2 AFTER I got the one for
1.4.41...
Thanks,
--Warren Selby, dCAP
On Apr 26, 2011, at 12:01 PM, Asterisk Development Team
asteriskt...@digium.com wrote:
The
On 04/26/2011 12:16 PM, Warren Selby wrote:
Which version of 1.4 is current, 1.4.41 or 1.4.40.2? I received both just now
and actually received the notification for 1.4.40.2 AFTER I got the one for
1.4.41...
Define 'current'. 1.4.41 is the most recent release from the 1.4 release
branch.
Hey folks,
I'm in Seattle for work on Thursdays and Fridays. Google doesn't know
of any UGs in Seattle, so I'm offering to start one. I will talk with
facilities about possibly hosting it at work, but if all else fails, we
can meet at Metrix Create Space or at a bar somewhere.
Let me know if
Hi,
I'm trying to get my head around an interesting problem (well I
think it's interesting :-) ).
An inbound call (say from extension 100) gets send to a queue and one of
the members (say on extension 200) answers the call when her/his phone
rings. In this case ${CALLERID(num)} = 100 and
Hi AM,
What should I do if I always want to ring a particular Queue member first
whenever he is available?
I don't think there is a Queue-strategy for that. What you could do is
have that person on a specific penalty (for example 10) and everybody
else on penalty 0. Before you do the queue
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