andrewgroup

While Emails from a PBX might Seem as useful as SNMP, I would argue that emails do not scale as well as an SNMP monitoring console. As a service provider we actively use WMI and SNMP to monitoring installed IT solutions and PBXnSIP. Expanding the use of SNMP is considerably easier than managing email alerts. An email alerting system requires 1 of 2 things A human being or an email parsing system. Needless to say the costs and complexiities of either are considerable and managing that can be a nightmare on VoIP Street. The fundemental problem with email fault notification is that we are forced to assume that all is well since no emails are arriving. "Out of Sight - Out of Mind." I can easily think of 50 things that commonly happen that would prevent those fault email alerts from arriving. But with a SNMP management consoles you easily define faults and alerts on failed SNMP queries as the first sign of a potential client trouble. As a service provider with SLA agreements with clients knowing on or before a client calls (on their cell phones) that a trouble may exist is a GREAT service. It would also be wonderful to get and provide additonal information that further solidifies the client relationship. Below is what we know about the SNMP variable and a few questions and a short wish list. these are the variables that we know of and my questions.. ;Call Objects Yes Calls (THE CALL OBJECT THAT MATTERS MOST IS TRUNK CALLS - Nobody cares about internal calls) ;Registrations Yes Registrations (WHAT ABOUT 2 PHONES REGISTERED TO SAME EXTENSION - WOULD THIS BE 2 REGISTRATIONS?) ;Messages Yes Minutes (THE MIB HAS THIS AS MESSAGES DOES THE TERM MINUTES FRO THE TEST MIB MEAN MINUTES WORTH OF MESSAGES?) ;Call Attempts No Calls (IS THIS VALUE DYNAMIC?) COULD THIS BE SET IN GLOBAL SETTINGS TO RESET EVERY 2 MINUTES? ;Successful Calls No Calls (COULD THIS BE CONTROLLED IN GLOBAL SETTINGS? ;Media CPU load Yes Value 0..100 (SELF EXPLANATORY) ;Successful Emails No Emails ;Unsuccessful Emails No Emails ;Email Alert Flag Yes Value 0..1 ;SIP Received Packets No Packets ;SIP Sent Packets No Packets Consider Adding ACTIVE TRUNK CALLS ACTIVE TRUNKS ACTIVE EXTENSIONS UPTIME Total Trunk calls, PER MINUTE, HOUR, DAY, WEEK, MONTH, YEAR (minute, hours and days would be enough, the rest is just a stretch) : - )

We will soon have a 10MB fiber SIP trunk with Always on full time recording. I'll be sure to report back with our successes and tears with any failures. :-)

Kudo's for Bills Post - He forgot to mention the oldest trick on the book for sharing lines "The Stick" http://multi-link.info/index.php?cPath=1

In post http://forum.pbxnsip.com/index.php?showtopic=1166 the reference is SSL is supported in V 3.0.xxxxx Using 3.0.0.2993 (Linux) and cannot make this work. Is this a hidden XML setting or are we overlooking the obvious? [8] 2009/01/15 00:20:11: SMTP: Received 220 2.0.0 Ready to start TLS [8] 2009/01/15 00:20:12: SMTP: Received [8] 2009/01/15 00:20:12: Last message repeated 2 times [5] 2009/01/15 00:20:12: SMTP: Connection refused on 74.43.63.17:25 [8] 2009/01/15 00:20:17: SMTP: Connect to 74.43.63.17:25

Has any thought or consideration been given to the idea of supporting SNMP trapping? This would create far less load on the PBX responding to GETs (not much I suspect) but would allow a far more proactive management by exception policy.

1. First be sure to add the trusted IP on the PORTS page on the system admin screens. 2. Reference the WIKI for OID info http://wiki.pbxnsip.com/index.php/SNMP 3. Download this easy - free wonderfully small non - intrusive SNMP GET - WALK tool from BPSoftware.Com (donate too) Presently tested and use Version 1.0.2.7 4. Start by simply copying the OID's from the WIKI into the OID Window and enter the IP address or the URL for your PBX. Note - Be sure to either add this .exe to your windows Firewall exceptions or disable Windows Firewall With simple tools you should be able to get your hands dirty using SNMP.... More info.....Download the SNMPUTIL from Microsoft or in their resource kits and use this next batch file to GET all WIKI variables. Perhaps we could have this copied and continuously updated by the development team to continuously keep us informed with changes in SNMP. ;sample batch file to get all WIKI variables ;Call Objects Yes Calls ;Registrations Yes Registrations ;Messages Yes Minutes ;Call Attempts No Calls ;Successful Calls No Calls ;Media CPU load Yes Value 0..100 ;Successful Emails No Emails ;Unsuccessful Emails No Emails ;Email Alert Flag Yes Value 0..1 ;SIP Received Packets No Packets ;SIP Sent Packets No Packets ; ; SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.1 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.2 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.3 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.4 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.5 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.6 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.7 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.8 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.9 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.10 SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.11 The above batch file results in the following outputs C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.1 Variable = .iso.org.dod.internet.private.enterprises.25060.1.1 Value = Integer32 0 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.2 Variable = .iso.org.dod.internet.private.enterprises.25060.1.2 Value = Integer32 14 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.3 Variable = .iso.org.dod.internet.private.enterprises.25060.1.3 Value = Integer32 41 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.4 Variable = .iso.org.dod.internet.private.enterprises.25060.1.4 Value = Integer32 28 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.5 Variable = .iso.org.dod.internet.private.enterprises.25060.1.5 Value = Integer32 224 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.6 Variable = .iso.org.dod.internet.private.enterprises.25060.1.6 Value = Integer32 0 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.7 Variable = .iso.org.dod.internet.private.enterprises.25060.1.7 Value = Integer32 99 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.8 Variable = .iso.org.dod.internet.private.enterprises.25060.1.8 Value = Integer32 0 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.9 Variable = .iso.org.dod.internet.private.enterprises.25060.1.9 Value = Integer32 0 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.10 Variable = .iso.org.dod.internet.private.enterprises.25060.1.10 Value = Integer32 349306 C:\SNMP>SNMPUTIL.EXE get pbx.yourdomain.com public .1.3.6.1.4.1.25060.1.11 Variable = .iso.org.dod.internet.private.enterprises.25060.1.11 Value = Integer32 349387

Running live on the test version ending in .3130 - Does this release contain more SNMP values suggested a PEAK call counter may be added.

Today we have a Snom 320 phone that refused to accept PNP auto with Buttons. remove the MAC address and you could manually set the phone up fine, place the MAC in the registration and shortly after booting you could not PING nor access the WEB of the phone. checked DHCP for odd reservation, moved phone to another switch, something was ODD. Trying to do as I preach by asking simple yes-no questions lead me to the PBX folder, assuming something was amiss I searched all XML files for the MAC address in question. Seems the MAC address of the phone that fails is in another extension config XML file in the extensions folder, I think the AVAIL field. We deleted and rebuilt that extension and VOILA' the phone would now autoprovision correctly.

I'm sorry, you've confused me with someone you might be paying for support. The least I can do is to do nothing and ignore your plea's for help. Instead, I've tried to point in the right direction. Best of luck to you. Cheers.

Did you not see the 200ms delays in your tracerts? 14 34 ms 33 ms 34 ms if-5-0.core2.NTO-NewYork.as6453.net [216.6.51.6] 15 34 ms 34 ms 34 ms if-1-0-0.core1.NTO-NewYork.as6453.net [216.6.97.14] 16 34 ms 33 ms 34 ms ix-8-0.core1.NTO-NewYork.as6453.net [216.6.82.42] 17 227 ms 228 ms 228 ms segment-124-7.sify.net [124.7.187.29] 18 227 ms 229 ms 227 ms segment-124-7.sify.net [124.7.187.5] 19 228 ms 228 ms 227 ms segment-124-7.sify.net [124.7.187.1] 20 227 ms 227 ms 228 ms 119.227.4.61 21 227 ms 229 ms 228 ms 210.18.5.141.sify.net [210.18.5.141] I could be next door and if my ISP or router injected 200ms+ of delay I would experience troubles. Tracert occurred at the bottom three layers of the stack and is application agnostic. You need to focus you attention on this issue. http://www.hinduonnet.com/seta/2002/04/18/...41800050300.htm

Importing accounts will need some additional functionality in the real world. With a soon to be real installation of 100+ accounts, we'll need to have these accounts members of predefined ACD and HUNT groups and many will have DID numbers. We'd like to import from an Excel file extension accounts with first, last, ext, aliases, etc ACD with aliases (as many as 100) 1st, 2nd, 3rd call stages, what extensions can join HUNT GROUP members and call stages What might make sense is the creation of a MASTER excel spreadsheet (Template) for use by all that can be imported by a dedicated function within PBXnSIP. We use a similar tool in our service desk software package. The developers created a series of EXCEL spreadsheets for project, client info, etc... Designing how a PBX should work is far easier on a peice of paper and this is something the client can sign off on, and once completed any changes can be made as a change request. ($$) changes cost money.

Sanity Check and commentary. We should soon be installing a new client with a dedicated 10Mb Fiber SIP trunk to deliver inbound 800 calls to 6 or more agent groups. 1 ACD may receive calls from as many as 100 800 numbers. Using Aliases in the ACD seems to be our first option. We might also consider adding more SIP trunks and having group of numbers spread across more trunks. Then using ERE's in the trunks to deliver calls to extensions, we would eliminate the aliases.. Before make the final decision, we want to better understand limitations, benefits in functionality or reporting. Using more SIP trunks would allow the system to scale as other servers can be installed on a departmental basis. Managing all of the alias in ACD's is going to be a pain in the #$%#$ if you wish to move or add a 800 number, but I guess that be problemation in the ERE expressions too.. The total 800 number count is near 500 and likely spread across 6 acd's. Having import function on ALIAS using CSV files might help in the future. otherwise COPY PASTE... Any Thoughts

Search Engines are a plus, asumming I'm looking for some advice and I have a specific topic of interest. Search engines are not very useful when you scan forums looking for interesting topics or to contribute an idea to another user.

Recently in our community Radio Ads are being run that depicts a scenario where nobody is donating blood, made more real with a final line saying, "What if everybody left giving blood to someone else?" This made me think of this and many other forums and the same though, "What if everyone who had potential answers left it to everyone else to answer or comment on questions?" We all know that if it has a Processor, Memory and Code, it isn't perfect and having access to people willing to provide free answers within these forum is a major plus. Please don't reply with any reference that PBXnSIP has an obligation to support their software, as they give away free short call licensing for testing and this puts the responsibility on the user. Microsoft Gives away 60 and 90 day trials of Server 2008, and I would not have free access to telephone support from Microsoft. I look in the forums of Shortel and others and I see few posts that appear to come from the development team. Given that I think a donation honor system (paypal) or otherwise for users of this forum to make a support donation that can be used in any way PBXnSIP decides - Beer - Pizza - Perks -Charity or to help improve PBXnSIP. As a VoIP provider in our City, our clients pay for our support services and we try to freely share our experiences with other users and in return we get tips too. Symbiotic - Chaotic relationship. I would also like to see two changes to the Forum Display. 1. A new Column Showing a Checkmark or something to indicate the issue has been solved 2. Changing the Name of the subject to better represent what the problem was. To often the subject line does not reflect what the final problem was found to be, and the content of the thread has a lot of good tips for all, but you wouldn't know from reading the subject nor do you know it was resolved. My .02 worth.

The way I see this is you have a US hosted PBX, with hope of extending US dialing to India. Right? Seems the final links on the journey are introducing 200ms plus delays - and we have found anything over 60ms has the potential to cause troubles. Several companies specialize in India VoIP service as we'll be doing the same this for a client in the next 3 months. Fix that problem. Our first attempt for this project will be with http://www.telifu.com Two Comments and a Wrap 1. Early you commented you had a Public IP, and now we hve a firewall. I would request a 1-1 NAT on the Public IP on the Firewall to your server. Opening RTP ports, adds processor and memory overhead to the SPI firewall and a 1:1 nat gets out of the way and passed all traffic far more effeciently. 2. For Everyone who reads this post, - this affirms the need to have a quality preinstallation plan - checklist. You can't built a square house on crooked bricks.

With an inbound call agent group set to record all incoming calls answers is their any reason that we cannot have a process scheduled to copy all of the recordings from this folder immediately? I see that Ad-HOC recordings are listed in the extension WEB access portal. Is this generated on the fly by reading the available ad-hoc recordings or does a database maintain a list of files? A database would likely be harmed by copying these AD-Hoc recordings. Does anything like this exist on the assigned recording options for EXTENSIONS, HUNT GROUPS or AGENT GROUPS?

Hopefully no more troubles pop up, but many of the questions I would be asking are below... A Server on a Public IP wouldn't have a firewall...............Correct? Did the Hosting Service present a package saying that they are VoIP - SIP Ready? We have setup all client premise based PBXnSIP systems regardless of Windows or Linux directly on a Public IP...another Internal IP serves the premise.... Have you disabled all unnecessary Windows Services? As I recall, some hosting services had either a software or hardware method allowing them to use cheap hardware to get the features of HP Lights Out Management. I'm not sure how that worked but I could the Hosting Service shed some light on possible problems caused by their management overhead? Is the Disk on this Server physical or part of an ISCSI shared disk space? I would want to know what lies between the Server and the phones - You can determine this using "trace routes" to identify various gateways and perhaps BGP routes out of the Hosting site. Can you put phones on another network (Eliminate your office ISP) and determine what the affects are? The question being asked by doing this is, "Does it improve when on another Network?"

I assume you meant to say, "The caller is not transferred after the present time expires." Can you make a HUNT group performed correctly? Are the Agent Group phones logged into the Group? Are you aware of the specialty flag for agent groups if no agents are logged in? An extension can be used as a mailbox - set DND - Phones registered to this extension should get WMI.

Let me share with you my basic rules of analysis - regardless of how complex the system. 1. Don't ask questions that cannot be answered yes or no. - If the answer isn't yes or no, simplify the question. 2. The goal of every question must be to determine those parts of the system that are working. Thus reducing the target area. The questions so far very high level questions and any answers are simply guesses. Let's start over.. Is this a VIRTUAL MACHINE in the hosting facility? Have you successfully configured and tested a Server 2003 in your offices?

Some time ago on version 2.0 and with a previous ITSP we had the following feature working fine. An outside caller would dial a DID and the call if redirected to the person's Cell Phone would see the Caller ID of the caller and not their office or extension. Somewhere in the middle of last year we upgraded that client to V3.x and that feature quit working, we didn't spend a great deal of time to address the trouble, but simply put them back on 2.x That client has since moved to another ITSP using BroadSoft V14 broadworks, and we upgraded them the the latest 3.x and that old problem reappeared. If we set trunk the remote party-ID (the calls are not forwarded) RFC3325 P asserted forwards the calls, but we see the trunk caller ID and not the original caller. The ITSP is quick to wash their hands and lay blame on PBXnSIP, but maybe it's just us and how we have set this up. We are exciting about partnering with the ITSP in our market, but need to overcome a few hurdles starting with this one.... Any help or feedback from someone on a Broadwords (CBEYOND) with this working as expected would be wonderful.... Cheers,

We've recently moved to a new ITSP that uses the BroadSoft Softwitch Platform - Where previously we were on a Cisco BTS10200 Previously this worked. Setting the Account Caller Redirection to forward calls to your cell phone and it displays the calling parties Caller ID vs. the Trunk ANI or your own extension. The Trunk Settings for Remote caller options are. Remote Call Party - ID - This fails to place the outboud call The Preferred and Asserted ID's forward the call but the ID is the Trunk ID I think we are educating the BroadSoft Administrators as SIP trunking is new to them. If you are working with an ITSP using BroadSoft and this is working for you, we'd like to know more..... Thanks

I also think a simpler solution is to add "1-" No quotes to the beginning negates the need for dial plan changes and keeps cell phone redirects fuctional. On a second note, since it detected the users CALLER-ID numbers, why force the PIN code and it seems this needs 10 digit dialing too.

We have been experimenting with IVR nodes to create simple screening. I.E. A Contract Client with extended support calls the office and presses 1 for support. It's in our extended period M-F 5:00pm to 7:00pm and we only want to accept calls from clients that know their last four digits of their phone number. I.E. The main autoattendent sent the caller that pressed 1 to the IVR.... The IVR message says, "Our offices are closed, Clients with extended hour coverage - please dial the last four digits of your office telephone number" We've experimented with several attempts to create and ERE to allow an "OR" exact match of two digits. Attempts like this are shown below. ![1234]|[6789]!300 I won't list all of my failings here and have reviewed every WIKI example and read WIKIPEDIA on the subject of ERE's but still no love. Can this be done and avoid the use of external SOAP?

On CS410 - Backup - Restore config makes this a breeze. In the Windows Worlds Folder Backup and SysPrep make for a good solution for production systems. I would recommend the trial key for the image. Manage your Dongle and NIC keys after the activation. The key to making this dependable is ZERO hardware changes allowing. However, we have experienced configuration file from older revisions of software have had some issues and every now and then you will have to refresh your master settings. Sometimes that's as simple as hitting the save button. Cheers,

Frederick - Rogier, Would you share information on the number of logged in users in the agent groups? What Phones are you using? We have soon to be approved project with 40+ sales agents logged into the same agent group and least busy ring will be the used. Agent Group Status, Info. Instant Messaging is Critical for the success of this project. Monthly inbound 800 numbers will peak near 200,000 minutes and missing calls is not an option. Are the troubles mentioned in this post firmly resolved and what advice might you give on managing these ACD Agent Groups? Cheers