what happened.
Dean
On Wed, May 4, 2011 at 2:31 AM, Johan Wilfer li...@jttech.se wrote:
On 2011-05-03 16:32, Dean Hoover wrote:
I am running Asterisk 1.16.2.13, dahdi 2.4.0 and libpri 1.4.11.4 on an
HP ML110 G6 using Ubuntu Linux 10.04 LTS.
I have two Digium TE121 single T1 port cards
/asterisk/messages. Still, only the most
serious errors are being reported to the messages log file.
It seems to work fine with my other Asterisk running 1.4.23.1. Is
there something else that I'm missing?
Dean Hoover
Waukesha, Wisconsin
That did it. Thanks everyone!
On Tue, Mar 29, 2011 at 9:11 AM, Andrew Latham lath...@gmail.com wrote:
On Tue, Mar 29, 2011 at 11:05 AM, Dean Hoover kb7...@gmail.com wrote:
I have an Asterisk server running 1.6.2.13, where I can't seem to get
the increased logging to save to the /var/log
an issue, and a complete shutdown of
the Asterisk server didn't change the results.
Any advice would be greatly appreciated.
Dean Hoover
Milwaukee, Wisconsin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
=yes
channel=96
---
On Mon, Feb 21, 2011 at 12:58 PM, Juan David Diaz juanch...@gmail.com wrote:
Dean,
what's your zaptel Zapata config_
regards
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 1:44 PM, Dean Hoover kb7...@gmail.com wrote:
We are running Asterisk
-
m, I would change the timing sources of the spans:
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
Have you try to plug the PRI into another Span that has been working
properly??
Regards.
Juan.
Linux User #441131
On Mon, Feb 21, 2011 at 2:23 PM, Dean Hoover kb7
On Mon, Feb 21, 2011 at 2:11 PM, Dean Hoover kb7...@gmail.com wrote:
Here you go:
/etc/zaptel.conf:
loadzone = us
defaultzone=us
span=1,0,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
#Added 2nd 2xT1 card
span=3,0,0,d4,ami
em=49-72
span=4,0,0,d4,ami
Source files aren't automatically installed on every install.
This link should help:
http://www.cyberciti.biz/faq/howto-install-kernel-headers-package/
Dean Hoover
Milwaukee, Wisconsin
On 9/27/2010 11:09 AM, Danny Dias wrote:
Hello,
I'm trying to compile DAHDI on DEBIAN but i have
on the equipment in our Florida office. The boss
doesn't want to pay it anymore, so I am right now moving them over to
Asterisk. ROI is less then two years.
--
Dean Hoover
Milwaukee, Wisconsin
--
_
-- Bandwidth and Colocation
in, let me know and I'll
post it when it's working.
--
Dean Hoover
Milwaukee, Wisconsin
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
(Cuba, etc) AND
that extension can't register from outside our network.
Using the default Asterisk settings is great for making sure that things
are working the way you want, but only after securing your Asterisk
server will it work the way you need.
Hope that helps. Good luck.
--
Dean Hoover
, then
# rotates the asterisk debug logs.
#
/usr/bin/find /var/log/asterisk -maxdepth 1 -type f -mtime +7 -exec rm
-rf {} \;
/usr/sbin/asterisk -rx logger rotate
exit 0
--
Dean Hoover
On 6/11/2010 11:08 AM, das sandesh wrote:
Hi All,
We have built an asterisk server (asterisk - 1.4.26.2) where
On 4/29/2010 3:10 PM, Jeff LaCoursiere wrote:
On Thu, 29 Apr 2010, David Backeberg wrote:
I'm considering a situation where I buy about twenty ATA devices.
I've played with the Linksys / Cisco PAP2T, and got that working fine
with some inbound and outbound faxing. The web GUI was okay.
On 3/8/2010 12:55 PM, Kevin P. Fleming wrote:
Dean Hoover wrote:
Our company has an Asterisk server where one of the T1 is connected to
an IVR. Asterisk is configured for FXO Loopstart, and the IVR is
configured FXS.
This is under control of the dialplan, though... using Dial(DAHDI/4
, but did not find anything. If you have a link that answers my
question, please send that.
SERVER CONFIG:
Asterisk 1.4.23.1
Zaptel 1.4.12.1
Digium TE210P dual-span T1/E1/J1 card
Thanks in advance,
--
Dean Hoover
Waukesha, Wisconsin
=Asterisk+sip+permit-deny-mask
Hope that helps.
Dean Hoover
Connor Spiess wrote:
Hi all,
I am wondering what people are doing for security when registering IP
phone’s remotely if you do not have the equipment to do a VPN tunnel at
the remote site. The phone I would be working with mainly
Warning: 399 Routing failed: ccbid=6361 socket=ast_server:5060
Confirm the Cisco CM can ping ast_server. Also try using the IP address
versus a name.
Dean
Jerry Geis wrote:
I am using asterisk 1.4.26.2 and cisco call manager 6.1
I am getting a 503 service unavailable message CCM.
Jerry,
I'm assuming that you mean that Cisco calls are making it to Asterisk,
but Asterisk calls are getting the rejection.
Looking at this again, there is probably not a network path issue since
packets are being sent and received.
Doing some quick research on what Cisco defines as a 503
Yes sir. Even upgraded to Cisco UC 7.0 and still works great with
Asterisk 1.4.23.
For me, I only had to change one thing. On the Cisco side, confirm the
SIP Trunk Security Profile has the Incoming Transport Type as TCP+UDP
and Outgoing Transport Type as UDP.
On the Asterisk side, a basic
We are upgrading to Asterisk as our company phone switch. We have our
own Auto-Attendant, call center and voice mail servers.
One issue that we are having right now is our voice mail server sending
Asterisk the NOTIFY to turn MWI on our Polycom SIP phones.
I see from the research I've done
that you have network access first, then verify that you didn't
make any changes to your configuration. If everything is good there, I
would work with your provider to make sure the settings are correct.
--
Dean Hoover
Network Administrator
Centurion, Inc.
262-317-5622 Phone
dhoo...@centonline.com
21 matches
Mail list logo
