andrewgroup

Over 3,900 views hit the Dynamic Service Flag post in Best Practices Forum since first Posted a year ago. There has been a very clear discussion and understanding of the need to have the ability to flip a scheduled flag. It's been 20+ years since I actually used Boolean and State Machine Logic, but isn't this a simple equation. Manual Flag A (MFA) Manual Flag B (MFB) Scheduled Flag A (SFA) AutoAttendant A (AAA) day Autoattendant B (AAB) night If MFA=0 and SFA=0 then AAA processes calls If MFA=0 and SFA=1 then AAB processes calls If MFA=0 and MFB=1 and SFA=1 then AAA processes calls If MFA=1 and SFA=0 then AAB processes calls If MFA=1 and SFA=1 then AAB Processes calls Just kidding..... The key is allowing designated extensions to simply flip a FLAG and allow the prescheduled timings to resume processing as occur..........

We are primarily using connections from ITSP that install a Local IAD Either a Cisco 2431FX8 or an ADTRAN Total Access 908..These have a 66 Block and the analog ports from these IAD's don't support the older centrex codes.The carriers in question will soon be offering true SIP trunks and will covert these clients to SIP. We have an 8 year history with NUVOX - the Old Gabriel Communications. They officially endorse SHORTEL, AVAYA, and a TRIXBOX install and we are pushing them as HARD as possible to allow us to ship them a PBXnSIP box for internal testing and approval on their list....... We got a paid support call and will gladly share the results.... Cheers,

"Life is a continuous humiliation of our pretensions." As you are well aware - if you told a client everything something doesn't do, we'd never make a sale, and I'm 90% sure whatever they would buy in our place, would have more limitations. I see more hours in or future..... Cheers

http://forum.pbxnsip.com/index.php?showtopic=665 been asking for this for over 1 year. Every small office PBX for the last 20+ years had a NIGHT button GREEN/RED for this fuction. I see several posts with dual service flags manual preceeding the auto.... Everythings a work-around.... Keep asking and perhaps V4.0 might have a direct solution to a old issue... Cheers,

Are any plans in place to allow the embedded OS to get various linux updates? The Apt-Get configuration is configured for a private Debian Mirror. It would also be nice to load a few more diagnostic tools such as IPTRAF, IPFM and perhaps some OS SNMP tools for DISK SPACE or CPU.

After several posts and 10 man hours, we have yet to get a dial plan that can send *67 (Cancel Caller ID" to the analog lines attached to the CS410... What would be better if you could assign a DIAL plan to the FEATURE CODE *67 so that the WIKI manuals match the capabilities. With these smaller installations, we are finding that we are once again becoming an engineering company vs. being a sales - support organization. Having proven dial plan solutions for the typical U.S. installation will go a long why towards moving more products and benefiting all users. For the Turnkey small business soutions, CS410's should come with a whole set of common Dial Plans. Wouldn't it be great if a ";" or something could be at the front of the dialplans to disable them allowing them to be commented out but still available.. Cheers, and still struggling to get *67 out the gateway..

I've attached a WORD document that contains a listing of what I think are the current set of Global parameters within the PBX.XML file.... Keep this file handy as a reference guide and a quick cut and paste when you need to flip a setting..... The plan is to break this file up and a denote on what setup page the setting may be accessible from also, but I'm not there yet. Enjoy

With 360 the default screen shows new and all messages. Is this available on the 320 handset... all latest V7.xx firmware?

When accessing the personal EXTENSION mailbox the default SAVE-AS for recordings "MESSAGE.WAV" needs some more flexibility. Perhaps selecting a DEFAULT save option in settings page. Consider Perhaps allowing INBOUND CALLER ID and/or CALLED NUMBER

This call log comes from a CS410 with 4 ANALOG lines and two active users. Also attached is the performance call graph. It's not possible that this system could make 6 calls. Is this a sampling error? No big Deal as we don't use or promote this as a useful graph, but knowing this exist might lead in subtle improvements.. 2/12/2009 15:52 USER1 (41) 13175321615 0:51 2/12/2009 16:02 Conf Room (43) 555533677 0:52 2/12/2009 16:03 PROP (3456743640) USER1 (41) 00:07 M 2/12/2009 16:05 BOISE,O (7676778887) USER2 (42) 2:55 2/12/2009 16:16 Conf Room (43) 66666676105 12:44 2/12/2009 16:35 COMPANY1 (1234566200) USER1 (41) 00:07 M 2/12/2009 16:35 COMPANY1 (1234566200) USER2 (42) 00:04 M 2/12/2009 17:05 CITY,OH (6565431783) USER1 (41) 00:15 M

from another PC run any of the free TFTP clients.....Run Wireshark in the process....TFTP lies dormant until a request is made... No Firewalls on the PBX please.

Setting BLOCK CALLER ID only works on SIP trunks. When Using an ANALOG gateway we need to dial *67 pause and then the dialed number to BLOCK the caller ID on this outbound call. What Methods have been created to Block POTS line Caller_ID on am FSO gateway? Would this be best to craft as a dial plan? Based on http://wiki.pbxnsip.com/index.php/Dial_Plan would the following suffice? \*([0-9]*) replacement is *\1

I've seen many posts with similar symptoms reported on CS410's and we have deployed more than a few over the last year or so and have never experienced any of these issues. Lucky perhaps, but here is how we have avoided these troubles. 1. We exclusively use SMART MANAGED SWITCHES and we take full advantage of all 802.1PQ tools and weighted Queues, and SNMP remote management. 2. We NEVER.. (Seldom) .....use a NATTED FIREWALL ahead of a CS410 or any PBX. When forced to do so for one REASON only.. The client has a SINGLE IP address. We specifically use an Intertex IX series router with SIP detection. We are presently reviewing an alternative for 1 reason only. Intertex does not support SNMP access or TRAPS. 3. ROUTERS FIREWALL ETC that have PNP autoconfigs enabled like residential LINKSYS, DLINK ETC gave a previous poster 3 weeks of really crappy performance. 4. DON'T Log anything unless it's in the act of diagnosing or Tweaking a feature. 5. USE SNMP wherever possible to gauge performance. 6. USE PBXnSIP PNP with all deployments 7. Know (Don't Assume) anything about the configuration of the analog lines by the telco provider and have the tools to verify the POTS lines are you hope they are. We see more and more POTS coming from IAD's and the config options for disconnect detections, ring volts and tones are many. REMEMBER the CS410 has a seperate 4 PORT ANALOG gateway in much the same way you would use an audiocode mp114 or other gateway.

First thought is, 24 extensions 10 thru 34 with service flags 50 thru 74 1 for each hour of the day, calls are redirected to cells based on the assigned call flags. Sounds like a maintenance nightmare after the installation to manage cell phone entrys. Perhaps a simple ACCESS web page that can SOAP update the cell phone numbers... Just a first thought...

We use "Clear Process" to create Call Flow Diagrams. Clients can review and sign off on the overall design and making changes is a SNAP via Drag-n-drop or text Scripting.. Once Approved - Changes cost $$

WITH a SIP open architecture, this is all possible, are you interested in a TurnKey solution and have a ready Budget? Are you the end user - or a reseller consultant?

Revisit an old POST http://forum.pbxnsip.com/index.php?showtopic=644

I think 7.1.35 gve us troubles with 1-way audio. We experienced 1 way audio, and a quick hold and unhold fixed this 1 way audio problem. We downgraded 1 release. This was late last summer or early fall. 7.3.14 is now on all phones.

first test you plans with 411, and move up from there, then to the not not so urgent 811 or whatever it may be, and to sound like and old Pro when 911 answers, say "Telephone Man (mam or sir), can you confirm the Emergency address and company name for this call?"

To further strengthen security only use switch technology supporting port security and the SRTP topic parallels one of the 10 rules of IT system security, "Your system is only secure as the people you trust."

No Love Moving to Dial Plan

No love - Moving to FEATURE request

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