search for: jmartin

...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/...

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

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

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

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...

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,

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

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

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

...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...

...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...

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'.

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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20071102/add66003/attachment.html

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

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,

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

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?

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

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