Hello,
I would like to use AMD on outgoing calls using analog line. I tested
with SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1
Other end is analog number behind another cisco/asterisk, also tested
calling a mobile number with the same result.
What I did: dial is done
Hello Experts,
I want to know if there is any way to modify welcome banner on asterisk console
when I connect using asterisk -r
Thanks,Haider --
_
-- Bandwidth and Colocation Provided by
You can do this
sip set debug ip x.x.x.x
On Wed, Mar 26, 2014 at 11:28 AM, Rafael dos Santos Saraiva
rafaels...@gmail.com wrote:
Hi
It's possible in Asterisk 1.8 enable verbose only in one context or
extension?
thanks
Att,
*Rafael dos Santos Saraiva*
Am 28.03.2014 10:32, schrieb Haider Khalil:
Hello Experts,
I want to know if there is any way to modify welcome banner on
asterisk console when I connect using asterisk -r
Hi,
did you compile asterisk from source? Take a look at main/asterisk.c
(line 174 in asterisk v 11.5.1). I think
Thank you so much everyone, I think I understand all I need for this.
Haider
To: asterisk-users@lists.digium.com
From: kevin.lar...@pioneerballoon.com
Date: Fri, 28 Mar 2014 11:18:16 -0500
Subject: Re: [asterisk-users] Asterisk CLI Banner
asterisk-users-boun...@lists.digium.com wrote on
Le 28/03/2014 15:40, Richard Mudgett a écrit :
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI
ad...@tootai.net mailto:ad...@tootai.net wrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I
tested with SPA3102 and cisco2811 as gw and asterisk
On Friday 28 Mar 2014, Haider Khalil wrote:
Thank you Thorsten Göllner.
Matthew,
What does violating license of Asterisk means ? Does it means I won't be
able to use any commercial modules or asterisk commercially ? I thought it
was open and anyone can change the code ? Haider
Nothing,
Modifying a program you have legitimately acquired is Fair Dealing.
The Law of the Land gives you the right to do that, even if the
vendor restricts your exercise of that right in practice by
withholding the Source Code.
That is false. Modifying a program is creating a derivative work.
As
Thank you Thorsten Göllner.
Matthew,
What does violating license of Asterisk means ? Does it means I won't be able
to use any commercial modules or asterisk commercially ? I thought it was open
and anyone can change the code ?
Haider
Date: Fri, 28 Mar 2014 10:00:42 -0500
From:
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI ad...@tootai.netwrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I tested with
SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other
end is analog number behind another cisco/asterisk, also
1) put a maximum number of loops in your IVR to terminate calls which are dead
or gone.
2) put maximum message length in voicemail.conf (ever tried to delete a 4 day
long voicemail?)
3) Call sometimes get stuck. This is life.
-Original Message-
From:
asterisk-users-boun...@lists.digium.com wrote on 03/28/2014 10:51:13 AM:
From: Haider Khalil haiderkha...@hotmail.com
Thank you Thorsten Göllner.
Matthew,
What does violating license of Asterisk means ? Does it means I
won't be able to use any commercial modules or asterisk
Richard Kenner wrote:
Modifying a program you have legitimately acquired is Fair Dealing.
The Law of the Land gives you the right to do that, even if the
vendor restricts your exercise of that right in practice by
withholding the Source Code.
That is false. Modifying a program is creating a
On Fri, Mar 28, 2014 at 9:10 AM, Thorsten Göllner t...@ovm-group.com wrote:
Am 28.03.2014 10:32, schrieb Haider Khalil:
Hello Experts,
I want to know if there is any way to modify welcome banner on asterisk
console when I connect using asterisk -r
Hi,
did you compile asterisk from
What does violating license of Asterisk means? Does it means I
won't be able to use any commercial modules or asterisk commercially?
I thought it was open and anyone can change the code?
Anyone *can* change the code. But it's licensed software, just like
most other software. The difference
Asterisk 11.1.0 running on Ubuntu 12.04 64 bit
Dahdi
Digium T1 card
Occasionally, I will find an inbound call that just seems to be stuck,
usually in our after-hours menu portion of the dial plan.
This morning I had this one
core show channels concise
Of course, any good attorney will never commit to anything. They
will never say it is alright to do X, unless X is do nothing
No, but a good attorney can give guidance as to likely expectations. As
you say, nobody can be sure of something even if it's previously been
established law, but a
On Fri, 28 Mar 2014, Richard Kenner wrote:
And this certainly may vary from jurisdiction to jurisdiction. For a
(quite dated at this point) discussion of this issue from a US
perspective, see
http://www.law.berkeley.edu/php-programs/faculty/facultyPubsPDF.php?facID=346pubID=157
The
Hello all,
I've got some PHP code that opens an AMI socket and does a ConfBridgeList
for a specific bridge ().
This all works just fine but I need to filter the information displayed to
only CallerIDName so I can see a complete list of names of participants.
After days of googling and
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.9.1
DAHDI-Tools-v2.9.1
dahdi-linux-complete-2.9.1+2.9.1
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
I (canadian) store has a deal on for the vera lite controller:
http://www.tigerdirect.ca/applications/searchtools/item-Details.asp?EdpNo=8930107sku=VEP-STARTER1
but this looks different than the vera lite green white:
?oops...wrong list :)
From: asterisk-users-boun...@lists.digium.com
asterisk-users-boun...@lists.digium.com on behalf of Michelle Dupuis
mdup...@ocg.ca
Sent: Friday, March 28, 2014 5:43 PM
To: Asterisk Users List
Subject: [asterisk-users] Best zwave controller
Why does the failed authentication place the number dialed, instead of the
username used, in the account field?
Any way to distinguish a failed dial attempt from a failed register attempt
using just the security log? (I couldn't see how looking at the log)
Hi,
I've setup TLS/SRTP with Asterisk 11.8.1 and wonder if there is a CLI
command to see if SRTP is active on a channel/call. I went through sip
show ... and core show channel... and did not see any mentioning of SRTP
while there is an SRTP call active.
Thanks,
Patrick
--
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398
I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I
believe this is related to the Makefile calling install_firmware with only 2
args, where
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote:
Unfortunately, after
http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1
2cc0661f3810ef47ad33206b2e398
I am unable to build DAHDI-Linux in a mock chroot for packaging
purposes. I believe this is
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