Displaying 19 results from an estimated 19 matches for "jmartin".
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2007 Feb 04
2
Roaming Profiles won't save
...Edition and followed the roaming
profiles example, but it still gives the errors.
One thing to note is the profiles live on the old Samba server and are being
mounted on the new server with NFS.
This is a small sample from the samba logs:
[2007/01/12 10:27:25, 2] smbd/open.c:open_file(352)
jmartin opened file jmartin/Desktop/prf11A.tmp read=Yes write=No (numopen=6)
[2007/01/12 10:27:25, 2] smbd/open.c:open_file(352)
jmartin opened file jmartin/Desktop/prf11B.tmp read=Yes write=No (numopen=7)
[2007/01/12 10:27:25, 2] smbd/close.c:close_normal_file(344)
jmartin closed file jmartin/Desktop/...
2011 Feb 10
2
Open Solaris on Sparc Virtual Server Options
Hello Everyone,
My question is simply what are my best options for successfully installing virtual tools on an Open Solaris OS running on Sparc Architecture. My hardware is a
Sun Blade 2500. From what I have read Virtualbox and Xen do not work on Sparc
technologies. What options do I have?
Thank you in advance for any advice,
Dan
--
This message posted from opensolaris.org
2011 Feb 11
2
Pull requests via Github - To Fork or Not? That is the Question.
Hi,
I''ve been coming up with some documentation examples, which I''d like
to contribute as Cucumber features to go into the relishapp.
I started following the contribute instructions on the rspec-dev
README[1] but have a question about pull requests.
Should I clone from the rspec/* projects on github, or fork them all
and then setup my local development environment from my own
2007 Jun 27
4
Using MSAccess to dial on a Zap line
Hello,
We use MS Access 2000 (I know, we're migrating away from it) as an application
to automatically dial phone numbers. The old phone system we have allowed the
call representative using the application to take their phone off hook, push
a button in the app, and the app would send the phone number to the phone
system and dial the number. We are moving to Asterisk for our main phone
2007 Sep 27
1
Zap channel stuck in conference
Hello, I have a strange problem with one of my Zap channels. A user told me
that he was in a voicemail system during a call, hit the Flash button, and
the call hung up. Now we get no dialtone on the phone hooked up to the
channel. Here's the status of the channel:
jmartin at rogue:~$ sudo asterisk -r -x "zap show channel 7"
Parsing /etc/asterisk/extconfig.conf
Channel: 7
File Descriptor: 27
Span: 1
Extension:
Dialing: no
Context: batoutbound
Caller ID: 5852311542
Calling TON: 0
Caller ID name: Metrix Matrix, Inc.
Destroy: 0
InAlarm: 0
Signalling Type: FXO...
2009 Jul 09
1
PRI failover to SIP trunk
Hello,
I've found a little documentation on voip-info and on the asterisk-
users list, although I was hoping for an example of a tried-and-true
failover setup between PRI and SIP.
We are an outgoing call center that uses asterisk 1.4 connected to 2
PRIs from the local telephone company in one group (g1) and a SIP
trunk from bandwidth.com. The PRIs are the primary outgoing service,
2009 Sep 03
1
Noises on Batphones
Hello,
The company I work for recently purchased 2 Rhino CB24s and a Rhino
PCI-E R4T1. The channel banks are plugged into the R4T1, as well as 2
PRIs from our telco. The CB24s are for all internal analog phones.
Most of the phones are setup in "batphone mode", which is
"immediate=on" in the DAHDI config. They are set up this way because
we are an outgoing call
2007 Sep 06
7
SIP Debugging to separate log file
Hello, I'm working with our SIP provider to nail down some call quality issues
we're having, and they've asked me to provide SIP debug log files from our
asterisk server. Is there a way to make asterisk 1.4 output only SIP
debugging to a specific log file? Or it is best just to use tcpdump?
Thank you!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY
2009 Dec 18
1
Could Asterisk be crashing under high context switches?
Hello!
I have been struggling with Asterisk 1.6 and DAHDI for the past few weeks. We are an outgoing call center with 30 internal analog phones hooked up to 2 Rhino CB24 channel banks. The banks are connected to a Rhino R4T1 card in a Dell 2950 server with 8 gigs of RAM. The 2 other ports on the R4T1 go to our 2 PRIs.
In this configuration, we have trouble maintaining stability. It may be fine
2009 Aug 18
3
Rules based on ipmasq
...and-squid-with-ntlm-auth-on-debian-etch.
Next to last point is firewall configuration by ipmasq but I have
installed shorewall. This is content of I89tproxy.rul file:
#!/bin/sh
#
# redirect http requests to non-local hosts to the transparent proxy
# GPL 2.0 or later (C) 2004 Johannes Martin <jmartin@notamusica.com>
# 1. do not redirect http requests to localhost
case $MASQMETHOD in
ipfwadm)
$IPFWADM -I -a accept -P tcp -D localhost 80
;;
ipchains)
$IPCHAINS -A input -p tcp -d localhost 80 -j ACCEPT
;;
netfilter)
$IPTABLES -A INPUT -p tcp -d localhost --dport 80 -j ACCEP...
2011 Feb 01
4
Delay Loading Rails Views
...om
Rails itself, not going through the Apache proxy. I think that
eliminates both the network and Apache as sources of the delay. I
think the delay is somewhere in Rails itself.
Here''s my logs. First, the commands and date/time stamps from the
prompt.
[litho:~/apps/timetrack/current/log] jmartin$ date ; curl -b
"cookie:val" http://127.0.0.1:10100/activity/list >/dev/null ;
date ;curl -b "cookie:val" http://127.0.0.1:10100/activity/list >/dev/
null ; date
Tuesday, February 1, 2011 2:28:38 PM GMT
% Total % Received % Xferd Average Speed Time Time
Time...
2005 Aug 02
0
Televantage 4
Hello,
I'm not sure if anyone can help, but I figured I'd ask anyway. I am
attempting to run the Televantage 4 Client under Wine 20041019 with
WineTools 2.1.2. It installs fine, but it does not run. If I make Wine
emulate win98, I get this error:
fixme:vxd:VXD_Open Unknown/unsupported VxD L"secprov.vxd". Try setting
Windows version to 'nt40' or 'win31'.
2007 Nov 02
1
(no subject)
Has anyone had any luck using 1.2beta2 on ps3. I was able to get 1.05
onto the ps3 but with the same code 1.2beta2 gives me only out of bounds
data after decoding.
--Justin
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2007 Jul 12
0
Additional Wildcard TDM2400P Setup
Hello,
We've purchased a 2nd Wildcard TDM2400P for our asterisk server. I've looked
around but I can't find documentation on how to properly setup 2 or more
telephony boards in asterisk. In particular, I want to ensure that the boards
are assigned the same device channels on boot (for example, if you have two
NICs in a Linux box and you want to make sure one card is always eth0
2007 Dec 05
3
No timezone in Voicemail email?
Hello,
I'm using Asterisk 1.4.14, and I've noticed that the emails that are sent out
when a user gets a voicemail don't have the timezone set in the header, so
they're appearing in the user's email clients at the wrong time. Has anyone
else seen this? I didn't find any bug reports or other info with Google.
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road,
2009 Sep 11
0
Asterisk 1.6.1.6 Crash when accessing Directory
Hello,
I may have found a serious issue with 1.6.1.6. I just compiled it
yesterday on our server. When anyone tries to access the name
directory through the Directory app, the asterisk process completely
dies.
Our extensions are in a realtime MySQL table, and the directory has
worked fine with previous versions of asterisk.
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Rd, Bldg 1
2007 Oct 25
2
Failover without shared storage
Hi,
is there any way to implement failnode pairs without shared storage? I
mean, two nodes with two syncronized separated hard disk and if one node
fails, the other one starts to work.
Thanks in avant,
Jos?
2007 May 14
1
Some problems with mysql CDR
Hello,
We have finally upgraded to Asterisk 1.4, however we've run into two issues
that weren't occurring before the upgrade.
Issue #1: We're an outgoing call center and need to record all calls. We use
the uniqueid field in the CDR to match with the recording, which we labeled
with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct
in the CDR. Here is the
2007 Sep 20
2
Outgoing SIP packets out of order?
Hello, I've been looking at some SIP packet dumps captured with tcpdump on the
PBX itself, and analyzed with Wireshark 0.99.4. I'm noticing something
strange, at least to me. All of the SIP packets going out from our Asterisk
PBX to either of our 2 VoIP providers are consistently 50% out of order. In
addition, if I use Wireshark's voip call player, the outgoing side of the
call