search for: pulaski

...c in Windows 2003 Active Directory. Global Group Maintenance is a member of Domain Local Group Maintenance. Domain Local Group Maintenance is afforded access to Linux directory /home/maint with this smb.conf share definition: [maintenance] comment = Maintenance Share valid users = "MYDOMAIN+Pulaski - Maintenance - DLoc" "MYDOMAIN+shawnadm" path = /home/maint writeable = yes create mode = 0660 directory mode = 0770 the directory: [root@pulaski-fs-001 home]# ll | grep maint drwxrwx--- 2 root MYDOMAIN+Domain Users 4096 Dec 15 13:11 maint getent group from the Samba box...

Ok, the Zhone sucks and the Adtran 750/850 seems to be a little too expensive. Can anyone recommend a decent channelbank that won't break the bank? TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

...eway and PBX (both *) are connected via 2xT1 (per-packet load balancing) between a Cisco 7206 and a 3640. When the total bandwidth pushes much past 50%, I start getting some crazy distrotion (jitter?), making it impossible for one or both parties to understand the other. TIA, -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638

Quick question: My PHP script is now able to connect to the manager port and successfully authenticate using MD5. I would strongly prefer not to do plaintext authentication at all. Would anyone object to plaintext authentication being left out? -- JustThe.net Internet & Multimedia Svcs. [The Fusion of Content & Connectivity] 22674 Motnocab Road * Apple Valley, CA 92307-1950 Steve

...rtner), or directly dialing an extension via an FXO port. Of course, I'd also end up with a few VOIP extensions as well. Does this sound like a feasable plan? Aside from eliminating the partner system (which isn't really an option), is there anything I'm missing? -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

...yet spoken to Mark about this yet. If I understood Chris correctly, he's going to work on finding a developer to write the drivers and get them committed to FreeBSD. I'm not up on Net/Open, so don't ask me about those, but I do want them included in this bounty. -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638 Pulaski Chamber 2002 Small Business Of The Year

...[<f897ebaf>] [<f897eccf>] [<f89c7852>] [<c01d0349>] [<c01d0568>] [<c01d2fa8>] Code: 8b 01 85 45 f0 75 1c 8b 02 89 d3 89 c2 0f 18 00 39 f3 75 e9 <0>Kernel panic: Aiee, killing interrupt handler! BTW, this is kernel 2.4.25-gentoo-r2 -- Troy Settle Pulaski Networks http://www.psknet.com 866.477.5638

The demo 1235 extension that Asterisk ships with works fine and the messages are sent to the address I set in voicemail.conf. I guess that means that my configuration is working perfectly so far. But when I set up another extension with a voicemailbox, no mail is sent when a message is left, although I can dial voicemail and listen to the message just fine which I guess rules out voicemailbox

When setting include => daytime|9:00-21:00|mo-fri|*|* How does this determine what is different between 9 AM and 9 PM And after hours ??? I want different hours on Saturday and Sunday And a different welcome message after hours Any help appreciated Regards Mick

Hi all ! Have anyone a resource / link for documentation to configure Asterisk to act as a SIP 2 PSTN gateway (ISDN PRI) ? Thx. Regads, Andreas. -- "If you want to pray. Go to the sea." ---------------------------------------------------------------- Andreas Czerniak <cognac@amcs.net> PGPkey http://pgp5.ai.mit.edu:11371/pks/lookup?op=get&search=0xEDB224EC

I have asterisk successfully dialing out using a X100P over a normal analogue PSTN line. But when I try to dial out over an analogue line that goes via a hardware PBX the call asterisk does not dial. Is there a configuration change I should make ? I am thinking of something like not wating for dial tone.

...answer, ring groups 1-3 for 30 seconds, and if still no answer, ring all 4 groups until the call is answered. What do I need to do to get this behavior? If the answer involves $$, tell me about it, I'm not afraid to spend some cash to help streamline my business. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 ~ 866.477.5638

...= 0, hard = 0) And, this user is definitely using some disk space: $ du -sx ~user3 796 /var/mailboxes/user3 Now, user2 did the same thing until I ran quotacheck, then it worked fine. Any ideas? Please respond directly, as I'm not subscribed to this list. Thanks, -- Troy Settle Pulaski Networks http://www.psknet.com 540.994.4254 - 866.477.5638

The new app_groupcount looks great for most applications but it a is a step back for call queueing... since app_queue calls physical interfaces and not extensions, app_groupcont can't be used to limit the calls passed to a dynamically added agent. I presently use the broken sip incominglimit feature (even though it's less than ideal as it also limits outgoing calls preventing

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

...owing operation" and subsequent facility IE containing the CNAM. BUT if you are seeing the CNAM on the TNT that may not be the issues, if this problem is present you usually will not see the name on the TNT either, just the number. > > Thanks, > > -- > Troy Settle > Pulaski Networks > 866.477.5638 > http://www.psknet.com > > > > Damon Estep wrote: > > If you are looking for real high density VOIP termination I would look > > at > > > >>something like a Lucent APX 8000, configure correctly it can pass > > &gt...

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

I would appreciate some help with this. I read the GPL license and basically it says you can do whatever you want with the software (sell, modify) as long as you include the source code, the License and make any changes you make available in the same manner to all others. My questions is this: If I develop an external application (say a Call Center application or a GUI management application)

I am sure this has been asked before, but why not use Google newsgroup or at least some forum BBS software instead of this cumbersome mailing list process? -- Costas Menico Meezon Software Corp 201-224-8111 costas@meezon.com --

Are they any issues still with hyperthreading processors, I've read and been told by a few people to make sure its disabled in bios if I want to use * on a hyperthreading machine. Kind Regards, Chris Bond