Hello
I am required to do some thing like Dail in modem .
User will have to call a modem just like we do in dail up connection
now we need to handle that request and retrieve some parameters
from that send a HTTp request to a web server and then after getting
http response send user a feed
On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote:
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set them to AUS zone
I have got this though
options wctdm24xxp
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set them to AUS zone
I have got this though
options wctdm24xxp opermode=AUSTRALIA
thanks
--
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On Thu, Jun 18, 2009 at 11:24:40PM -0500, Karl Fife wrote:
After a kernel update (but before rebooting) Is there a way to recompile
Zap/Dahdi against the new kernel?
My objective is to eliminate the additional downtime that occurs while
recompiling/installing zap/dahdi after booting into
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco
7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem
is that Cisco phone isn't authenticated on Asterisk.
In tftp directory I have:
apps41.1-1-1-15.sbn
cnu41.3-1-1-15.sbn
copstart.sh
Hi,
Is there a way on Polycom phones to show an agent whether he is logged
in or not?
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On Fri, Jun 19, 2009 at 11:08:49AM +0300, Tzafrir Cohen wrote:
On Fri, Jun 19, 2009 at 06:04:17PM +1000, Alex Samad wrote:
Hi
I am seeing this in my syslog
[235900.797660] dahdi: Registered tone zone 0 (United States / North
America)
I am in Australia so I would want to set
From memory, it is doable but this is a feature that Polycom never quite
finished writing.
PaulH
On Fri, 2009-06-19 at 10:58 +0200, Louis-David Mitterrand wrote:
Hi,
Is there a way on Polycom phones to show an agent whether he is logged
in or not?
Hi Moy,
I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI
patch, which fixes a bug about stopping AsyncAGI applications, as may be
you can recall from the thread [asterisk-users] async agi question in
Well, at least this did not add to the wait time of your callers :)
It should be possible to do silence detection/removal automagically using
sox as well - see e.g.
http://www.justlinux.com/forum/showthread.php?t=136678
2009/6/18 Louis-David Mitterrand
I fail to see how this script is useful in order to use Snom's
Plug'n'play config.
Who said it does?
The Topic is snom mass deploy - not Plug'n'play config.
It does not use snoms Plug'n'play config, but it still provides for
snom mass deploy using the phones' built-in dhcp/http mechanism.
Nir Simionovich is about to become a father. He will be joining our
conference at 12 Noon EDT today from the Maternity Ward to talk about
Amazon EC2 cloud computing with Asterisk. Nir gave a very good
presentation on this at AMOOCON a few weeks ago (see
http://www.amoocon.de for more on that). The
Conrad Wood wrote:
On Thu, 2009-06-18 at 14:21 +0200, Philipp Kempgen wrote:
On Jun 18, 2009, at 7:25 AM, Alex Samad a...@samad.com.au wrote:
I am trying to setup asterisk to do a mass deploy of some snom
phones. I
can't find where i configure asteriks to listen to the
On Thu, 2009-06-18 at 19:18 -0400, John A. Sullivan III wrote:
On Thu, 2009-06-18 at 14:55 +0200, Giorgio Incantalupo wrote:
Hi John,
I already have the ccd dir with the iroute (mandatory for routing to
pc/phone connected to vpn client). During the last test I could register
and
Alex Samad schrieb:
On Thu, Jun 18, 2009 at 02:21:47PM +0200, Philipp Kempgen wrote:
Snom supports what they call PnP config.
Technically:
---cut---
# SIP Event Notification:
# http://tools.ietf.org/html/rfc3265
# SIP UA Profile Event Package:
#
On Fri, 2009-06-19 at 11:45 +0500, ABBAS SHAKEEL wrote:
I am required to do some thing like Dail in modem .
User will have to call a modem just like we do in dail up connection
now we need to handle that request and retrieve some parameters
from that send a HTTp request to a web server
On Fri, Jun 19, 2009 at 5:32 AM, Jose Ariascyr2...@gmail.com wrote:
Hi Moy,
I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI
patch, which fixes a bug about stopping AsyncAGI applications, as may be you
can recall from the thread [asterisk-users] async agi question in
Sasa wrote:
Hi, I use Asterisk-1.4.22-3 (on Trixbox) and I have a problem with Cisco
7941G with firmware SIP41.8-0-2SR1S (but also with SIP41.8-3-1S), my problem
is that Cisco phone isn't authenticated on Asterisk.
In tftp directory I have:
apps41.1-1-1-15.sbn
cnu41.3-1-1-15.sbn
Can't tell you the how, but you should be able to do this as a BLF or buddy
function perhaps using hints. I know the GUI can tell if an agent is logged
in or out, so it can't be that hard.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
I've found that different types of TFTP servers return differing errors when a
file doesn't exist. You don't need the TLV file, but you do need a distro that
tells the phone it's not there correctly. I have not had ANY luck with windows
tftp servers, only linux.
-Dave
-Original
The free solar winds TFTP server worked well for me, as well as the
CentOS TFTP server
The Solar Winds one produces an on screen log file which is very nice
while troubleshooting
The Cisco 7960's I have set up want to find the file name, but seem not
to care if it is empty.
Both with the
John Novack wrote:
I have had sucess with creating a zero length file named
CTLSEPmac_address.tlv
Or whatever the damn thing wants, and it then seems to be happy.
With Cisco 7960's
Your results may vary
...with CTLSEPmac_address.tlv in tftp dir in log file I have:
Using local port 3131
David Gibbons wrote:
I've found that different types of TFTP servers return differing errors
when a file doesn't exist. You don't need the TLV file, but you do need a
distro that tells the phone it's not there correctly. I have not had ANY
luck with windows tftp servers, only linux.
I have
When I try to use 1.6.1.1 with ODBC and MySQL, I get these:
[Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table
supporten_...@asterisk: column type (-9) unrecognized for column 'name'
[Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table
supporten_...@asterisk: column
On Fri, Jun 19, 2009 at 05:25:18PM +0200, Sasa wrote:
David Gibbons wrote:
I've found that different types of TFTP servers return differing errors
when a file doesn't exist. You don't need the TLV file, but you do need a
distro that tells the phone it's not there correctly. I have not had
On Friday 19 June 2009 10:25:15 Benny Amorsen wrote:
When I try to use 1.6.1.1 with ODBC and MySQL, I get these:
[Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table
supporten_...@asterisk: column type (-9) unrecognized for column 'name'
[Jun 19 17:19:22] WARNING[5882]
What does your SEPMacAddress.cnf.xml file look like? In my experience,
the XMLDefault.cnf.xml file is not retrieved from my 79x1 devices and I had
to specify the firmware version in each SEP file. I am using 8-4-4S, but
for you this would be something like this:
device
Hello
Actually i am required to make two application
1) that user use
2) that is deployed on server
Application for user will be just like the windows standard connection
using dail up modem but user will dail my PSTN number instead of the
number we inter provided by ISP.
on deployed server
is it just me or am i right in thinking this has nothing to do with
asterisk?
2009/6/19 ABBAS SHAKEEL shakeel.abbas@gmail.com
Hello
Actually i am required to make two application
1) that user use
2) that is deployed on server
Application for user will be just like the windows
Geraint lee
I also dont know .what kind of requirements are these :P
i am just looking if it can happen
On Fri, Jun 19, 2009 at 9:33 PM, Geraint Leegera...@gmail.com wrote:
is it just me or am i right in thinking this has nothing to do with
asterisk?
2009/6/19 ABBAS SHAKEEL
On 19 Jun 2009, at 17:33, Geraint Lee wrote:
is it just me or am i right in thinking this has nothing to do with
asterisk?
My thoughts too. Was keeping quiet incase I was misunderstanding.
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Steve Totaro wrote:
On Thu, Jun 18, 2009 at 12:46 PM, Brent Davidson
br...@texascountrytitle.com mailto:br...@texascountrytitle.com wrote:
John A. Sullivan III wrote:
Hello, all. I am delightfully slogging my way through installing and
configuring Asterisk 1.6.1.1 on CentOS
What are the HA options for Switchvox systems?
Is it possible to set up redundant systems with DRBD?
I know on the digium website they talk about Optional cold spare
failover What does this mean? Is this an active spare ready for some
sort of automated failover?
Thanks for you help,
Bob
Hi Danny;
I found cfgbasic.html under the /var/lib/asterisk/static-http/config and did
not find cfgadvanced.html, any advise?
About the root directory: do u mean that I have to set my root directoty to be
/var/lib/asterisk/ at the httpd server? Because by default the httpd server has
another
On Fri, Jun 19, 2009 at 10:28:28AM -0700, bilal ghayyad wrote:
Hi Danny;
I found cfgbasic.html under the /var/lib/asterisk/static-http/config
and did not find cfgadvanced.html, any advise?
cfgbasic.html is now merely a redirection to index.html .
cfgadvanced.html is now gone - the advanced
On Fri, Jun 19, 2009 at 10:28:28AM -0700, bilal ghayyad wrote:
About the root directory: do u mean that I have to set my root
directoty to be /var/lib/asterisk/ at the httpd server? Because by
default the httpd server has another root directory than this, or you
are talking about another
bilal ghayyad schrieb:
what about the port 8088, from where I can set it (in case I need to change
that port to be another port)?
That would be the bindport parameter in /etc/asterisk/http.conf
I guess.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -
why is CROSS_ARCH=Linux? is this something the AVR32 distro is doing, or
something you did? it should be something line avr or avr32
On Thu, Jun 18, 2009 at 3:08 AM, Paulo Santos paulo.r.san...@sapo.ptwrote:
Greetings everyone,
I'm trying to compile asterisk for an AVR32 (Atmel NGW100).
Hi Steve
I tried your script :
STATUS=$(sudo asterisk -rnx pri show span 1\
| awk '/Status/ {print $3}'\
)
if [ Up, == ${STATUS} ]
thenecho PRI UP
exit 0
elseecho PRI DOWN
exit 2
Do you need to path sudo (/usr/sbin/sudo)? Try running the script as nobody
and see what happens.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sriram
Sent: Friday, June 19, 2009 1:21 PM
To:
On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote:
In modules.conf: noload = cdr_csv.so
Are there other modules I need to load or unload ??
asterisk*CLI module show like cdr
Module Description
Use Count
cdr_addon_mysql.so MySQL CDR Backend
0
For determining security risks, its specific to how your dialplan is set up.
If a person connects to your asterisk, what can they do? what happens? did
you set the incoming context to one with outgoing dialing rules?
Also for filtering calls, you'll probably want to either look at the
incoming sip
Hello, all. I am attempting to use IMAP voice mail storage in Asterisk
1.6.1.1 on CentOS 5.3 using Zimbra 5.1.6. I will not be using it as it
has proved terribly unstable - Asterisk segfaults on every voice mail
message although the message is successfully deliver to my email inbox -
but I
jonas kellens escribió:
On Thu, 2009-06-18 at 11:52 -0500, Tilghman Lesher wrote:
In modules.conf: noload = cdr_csv.so
Are there other modules I need to load or unload ??
asterisk*CLI module show like cdr
Module
Description Use
John A. Sullivan III schrieb:
By the way, how does one disable IMAP storage?
I did not see a module for IMAP storage. It would seem strange that I
would have to recompile.
Sadly you have to recompile.
Disable voicemail IMAP storage in `make menuselect`.
Philipp Kempgen
--
AMOOMA GmbH
On Fri, 19 Jun 2009, Sriram wrote:
I tried your script :
STATUS=$(sudo asterisk -rnx pri show span 1\
| awk '/Status/ {print $3}'\
)
if [ Up, == ${STATUS} ]
thenecho PRI UP
exit 0
elseecho PRI
On Fri, Jun 19, 2009 at 1:02 PM, Bob Pierce pier...@westmancom.com wrote:
What are the HA options for Switchvox systems?
Is it possible to set up redundant systems with DRBD?
I know on the digium website they talk about Optional cold spare
failover What does this mean? Is this an active
Hi on the list,
does anyone of you have experience with asterisk 1.6 and mISDN, pri
primarily?
Thanks in advance Regards,
Christophorus
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We're using Asterisk 1.6.1. When our SIP clients have silence
suppression turned on, it's a problem for many apps. Is there a
workaround for this in Asterisk? Other than turning silence
suppression off in the SIP client, is there anything I can do on the
Asterisk side to make things work
Just turn CNG on the phone and it should be fine ;-)
On Fri, Jun 19, 2009 at 6:08 PM, Bryan Field-Elliot
bryan+asterisk-us...@nextalarm.com bryan%2basterisk-us...@nextalarm.comwrote:
We're using Asterisk 1.6.1. When our SIP clients have silence
suppression turned on, it's a problem for many
I have an Asterisknow.org CD. When I boot up, it seems ready for me to
choose update, console, etc. I'm assuming I need to do something at the
CLI prompt. Is there a tutorial that would take me from loading CD to
making first test call?
Computer is Dell Optiplex GX260
50GB free disk space
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