andrewgroup
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Posts posted by andrewgroup
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well for the sake of this post, and in an ideal scenario in the future everyone PBXnSIP system is attached to an ISP provider that fully supports QOS and this isn't an issue. So for the sake of following a best practice, what are the must do's to accomplish the goal of business to business calling. Perhaps some links to the state of Enum might help explain the reference to ENUM, Long story..
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I've seen many vendors in the SIP / VOiP industry with business cards showing a SIP telephone number..
301@sip.domain.com
could someone explain how this is to work in PBXnSIP?
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For the sake of this post, let's assume that at some point every PBXnSIP deployment will be connected to a Static IP address and can or will be resolved at SIP.newco.com
The client will want to be able to accept incoming calls via their normal TDM carrier via POTS, PRI gateways or VIA an ITSP using SIP TRunks, but also would like to accept alternate calls via ext@sip.newco.com or the future using ENUM.
what might be a standard list of requirements to meet this challenge?
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Many small businesses wish to take advantage of Automated attendants want the following features, all of which consume accounts..
1. A daily Greeting (This can be scheduled based upon a known Holiday Schedule using a scheduled Service Flag
2. A lunch Hour Greeting (This Can be scheduled based uipon a known Holiday Schedule)
3. An AfterHours Greeting (This Can be Scheduled based upon a known Holiday Schedule)
Of course the client also wishes to be able to turn a greeting on at any time for whatever reason, (Late or Long Lunch)
or turn on the Offices are closed due to weather.
I'd like to hear how others have addressed these issues to enable the client to control these and eliminate busy work to make the client happy on these issues?
What about remotely accessing and enabling these toggle?
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Based upon our experiences and what I read within this forum, I think it would be appropriate and helpful if a Best Practices Forum was created. The technical features of PBXnSIP are many and when left for everyone to figure out, the solutions and setups of these are going to very greatly.
Based upon things learned in training, we are going to redeploy our installations before doing more. Establishing some basic guidelines early will make things easier in the future.
Standard Extension Numbers
Standard AutoAttendent Numbers
Standard Service Flag Numbering and usage
Using internal FQDN vs. IP address Registrations
Using ENUM DNS recommendations
End User TIPs sheets
there is just so much to consider, and a wealth of questions in the forums all suggesting that some standard best practices should be adopted by all in order to make this a better product.
Maybe it gets started by creating the Best Practices Forum and we as a group post things that we are all doing, not as an answer to a question, but as a potential tip to others.
Cheers,
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Well, it can be done. See http://wiki.pbxnsip.com/index.php/Snom. Just make sure that you are using version 7.1.28 or higher. There are images available at http://www.pbxnsip.com/protect/snom3[0267]0-7.1.28-SIP-f.bin, just in case that you don't find them on the Internet.
The monitoring of the DND button for other extensions might be a little challenge, though. Maybe this is something that we need to add...
The last sentence is the crux of the issue, Monitoring the status of another extension (The Boss) for the receptionist was the original post. We are also trying to avoid too many client exposures to upgrade firmware on 50 phones. We are waiting until SNOM addresses a linger problem of connecting to inbound callers together. This has been an issue for the last 12 months on all firmwares.
Try this on PBXnSIP with SNOM 320 or 360 phones, accept an inbound call. place that caller on hold. Answer another inbound caller - simply hang up on the second caller and see what you get. My guess is the two callers are now talking to one another. (Two upset clients can have a real hay day talking about your phone system. So until we can deliver that fix to several clients we are strategically avoiding making any promises and wasting ours and their times when we aren't resolving any of the bigger problems. Oh, and you can turn the advanced snom setting Call join on Xfer (2 calls): on or off, it really doesn't matter.
If I were technically capable, I think adding some SNOM action URL's to the PBXnSIP web server to do some cool stuff like this would be a real plus. The SNOM web - wiki speaks of this exact example, but we simply don't want to go about creating 1-off solutions.
Cheers.
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If the correct answer was "It can't be done" that would be sufficient, however, I was hoping that I might spark a creative use of the SNOM advanced action keys for PRESS DND, I'm not capable of creating the solution, but until the idea of having a second status on extension monitor key like blink for DND is considered, maybe when pressing DND triggers an action that triggers some sort of alert to other SNOM phones. Perhaps a scrolling message on the SNOM phones
Since I have an existing 50 station PBXnSIP spread in two floors of 12,000 sq. ft. and the front receptionist not knowing if the Boss has his DND button is causing us some embarrassment, we got it covered, but If I were selling a PBXnSIP competitor, I'd be reading these forums and discovering these little jewels for my competitive analysis. I'm sure purchasing decisions have been made on less.
Cheers,
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With two Clients using V2.xx using SNOM 360/320 phones, the receptionist needs a basic feature to duplicate the functionality of 20 years worth of user training with almost every TDM phone system before the SIP/VOiP Era..
The Receptionist has a programmed button / light for the BOSS....
The Boss pressed DND or the * CODE DND
But the receptionist has no way to know not to send a CALL to the BOSS...
the RECEPTIONIST has no computer,
Why CAN'T PBXnSIP blink a like on monitored stations with they are put in DND via the button or the STAR Code.....
For all the great features, not being able to provide the receptionist DND status for the BOSS and others is a glaring hole and we get bugged for this weekly..
Did I miss something or is this possible?
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How does one assign an alias to the one existing localhost?
Based on recent knowledge gained at a recent PBXnSIP training class, there are long term benefits to have all domains named to match what might become a FQDN to accept true SIP calls, not just calls from an ITSP...
Please tell me alias can be added ro an existing localhost domain.
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The reasons to do this are many. Let me explain how this bug embarrased me today. I had new client prospect on my first call and I conferenced in the customers telecom vendor. We planned to speak to that person directly but if we reached voice mail we were going to leave a joint callback message. I began hitting the proper extension numbers to access the person and my prospective PBXnSIP client began hearing the continous tones. PBXnSIP timed and then hung up on the caller.
We can duplicate this condition 100% of the time and would have no problem extending our system to help diagnose this condition.
In this case, the PBXnSIP condition demonstrated itself to a prospective PBXnSIP client and I had to make an excuse for the trouble.
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A dedicated section to identify known bugs and work-arounds that we should all be aware of. These could simply be copies of other posts that might have been posted and and have been confirmed as bugs due to be addressed in a forthcoming release or patch.
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When setting up a Cell phone to ring immediately or delays on an extension, it appears to send the callerID of the PBXnSIP box to the cell phone. Is it possible to send the original callers ID. (This might be asking to much)
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SIP Registrations - success or failures
Current SIP Call Paths
Total SIP registered devices
Gateway Device Registrations (customizable) Analog, PRI etc)
Current Call Count
5 Minute Rolling Average Call counts
Disconnected Calls (if possible)
Failed Administrator Login attempts.
Successful Administrator Login Attempts.
Conference Room Activity.
Conference Registration.
Total Voice Mail Message counts.
Total Minutes used.
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I couldn't agree more. We are new to pbxnsip, and the single thing spend the most time on is translating technical notes into how to use the features. A list of features is far cry from a users manual. If pbxnsip stays true to the channel, then these documents need to be available for customization by us. If anyone knows how to make fonts, it would be nice to have a font for the buttons on the popular phones. Snom, Cisco, Poly and others... I'll contrib, anyone else?
Interoffice Trunk Setup with 3 PBX's
in Trunk Setup
Posted
Building SIP trunks between 3 offices allowing direct dialing extension
Office 1 uses extension 100 to 200 same for the 2nd and 3rd office..
would creating interoffice SIP trunks to allow direct dialing VIA sip using 5 digit dial with the 1st 2 digits being the office code be the most affective way to accomplish this?
assuming the 2 digit offce codes are 50, 51, and 52 what would the dial plan look like to route the call and to recieve a call?