Asterisk 1.4
We are experiencing an issue on transfers where no progress tones are heard
by the caller:
1. Call from 1593 (SPA525G 0026998D2) to 1595 (SPA922 000B820AF).
1595 answers
2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4A). 1595 hears
MoH.
3. 1597 starts ringing and
On 06/05/2012 12:39 AM, Klaverstyn, David C wrote:
Hi Guys,
All my installs are based on PRI ISDN. I now have a site that I need to
install BRI. As I have not done a BRI install before I’m wanting to get
some information from the people in the know if I need to do anything
special.
Typically I
hello,
is there someone who successfully get info about attended transfer from CEL?
if yes, can you post some hints/algorithm/...?
scenario
A - customer
B - secretary
C - consultant1
D - consultant2
A - B
B axfer C
C axfer D
i need to know
time B with C (consultation)
time A with C
time C
When will this be available at packages.asterisk.org?
Thanks!
EKG
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We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
to grow to 300MB in just 5 minutes, which eventually overloads the
Hello guys,
I need to be able to throw cdrs on more than one servers at a time. Please let
me know how this can be done.
Thanks
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Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Owais Ahmad
Sent: Tuesday, June 05, 2012 7:54 PM
To: Asterisk Users
On Tue, 5 Jun 2012, Owais Ahmad wrote:
I need to be able to throw cdrs on more than one servers at a time.
Do you mean:
1) I need to collect the CDRs from multiple servers in one place so I can
report on them.
Or
2) I need to distribute the CDRs to multiple servers so I won't lose
During testing with the Digium phones I have run into a problem where I try to
make a change to the sip device name. I make the device name change in sip.conf
then make the matching change to the lines in res_digium_phone.conf. I then do
'sip reload' and
'module reload res_digium_phone.so'. I
On 06/05/2012 10:23 AM, Chet W. Stevens wrote:
During testing with the Digium phones I have run into a problem where I try to
make a change to the sip device name. I make the device name change in
sip.conf
then make the matching change to the lines in res_digium_phone.conf. I then do
'sip
Yesterday a customer was attacked from the following IP addresses so
add them to your blacklist:
iptables -A INPUT -s 37.8.119.75 -j DROP
iptables -A INPUT -s 37.8.22.240 -j DROP
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Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
We use complete regional blocks from Wizcraft and blocking at minimum
all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We
block almost anything that is not our actual customer market and screw
the rest.
On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavez cur...@telecomabmex.com wrote:
I figured out the problem. Actually the sending fax machine speed was set
as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully
set to use T.38 protocol. After that I was able to receive fax.
Thanks Tim for assisting me out :).
- Original Message -
Hi Tim,
I'm
On Tue, Jun 5, 2012 at 12:47 PM, Ahmed Munir ahmedmunir...@gmail.comwrote:
I figured out the problem. Actually the sending fax machine speed was set
as 33000 bps, later I set to 14400 bps and in my dial plan, I forcefully
set to use T.38 protocol. After that I was able to receive fax.
Thanks
I am trying to figure out the best way to deal with this. I want all of the
calls in the network to be G729 and this is working. I do have hardware
that provides me 30 g729 licenses. I am setting each extensions to
disallow=all and allow=g729. However when I have this setup, I get no voice
mail
What does the output of g729 show licenses show? If it doesn't show licenses
then Asterisk is not licensed for G729 codec.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
Sent: Tuesday, June 05,
The G729 is coming from a Sangoma D100-030 card and the G729 transcoding is
working, the only issue I have is not hearing prompts from the system.
On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling ewiel...@nyigc.com wrote:
What does the output of g729 show licenses show? If it doesn't show
When you installed asterisk, did you also enable the installation of the
g729 asterisk-(foo)-sounds options in 'make menuconfig'?
On 6/5/2012 2:46 PM, Tim King wrote:
The G729 is coming from a Sangoma D100-030 card and the G729
transcoding is working, the only issue I have is not hearing
Can you give me some pointers on where to read documentation on how to
set up registered phones?
Also I'm wondering if maybe it would help if I tried setting up some
softphones first.
Can someone recommend some cheap softphones that work with asterisk?
Jacob
On Tue, May 29, 2012 at 5:36 PM,
Hello,
I know this a little bit off topic but maybe someone of you has experience
with this topic.
I want to run a hylafax install in an openvz container with iptables.
My Problem is that I can load the kernel module for ftp connection tracking
(nf_conntrack_ftp) in the containers config file
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