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CURRENT SETUP We are using a traditional phone company (comcast - I guess you can call them traditional). They offer a type of hunt group which allows the calls that come into our phone number to ring the first line and then after 20 seconds go to voicemail or if the first line is busy, ring the second line for 20 seconds and then go to voicemail or if the seconds line is busy ring the third line for 20 seconds and then go to voicemail. If all lines are busy just go to voicemail. We have a ring group setup on the panasonic kx-td 1232 so that the office phones ring if any calls come into any of the 3 lines and the call gets connected to whichever phone picks up first.
NEW PROVIDER We are trying to port our number to a SIP provider (VoIP.ms) - They only offer call hunt which rings the first line for 20 seonds then rings the second line for 20 seconds and then rings the third line for 20 seconds, then goes to voicemail. This makes very little sense because people will have to wait with the call ringing for 3 x 20 seconds just to get voicemail if nobody answers any of the lines. We have a ATA/FXO box (Cisco SPA8000) with 8 analog FXO ports on it.
MULTI-RING / SIMUL-RING The new provider also offers ring multiple lines at once, So I thought about just making it so each extension in the office was assigned to a separate line (for example extension 101 would ring immediately on CO1 only and extension 102 would ring immediately on CO2 only, etc...) then I would have the new provider ring all lines at once and all 8 of the office phones would light up and ring for 20 seconds and the first call to grab it would connect the call automatically. The issue we have now is that it takes a moment for the PBX to realize that the call has stopped ringing so everyones phones keep ringing after the call has already been picked up and so a number of other people try and pick up the call and just get a dial tone.
OTHER OPTIONS? Is there any other way to setup this system to allow compatibility with new phone providers for VoIP and SIP calls. None of the sip providers seem to support traditional call hunt round robin or whatever its called. My experience is that most old analog PBX's are setup to support this older type of call handling.
Any ideas? thanks for any feedback.
-Jemmyn [Panasonic KX-TD1232]
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I can’t see why they have to hunt through the 3 lines before voicemail. Especially if it is sip, even going through ata’s
With the 1232 you don’t have sip option, you are still working with analogue lines at the system side. Normally old lines would only roll over if the other one is busy. If the line is ringing on line 1, what use is it going to line 2 and act as a second call on the system
I would upgrade to NS and connect via sip directly and not use the Ata adapter or get a voicemail unit for the 1232 and don’t use the providers mailbox
“I have not failed. I've just found 10,000 ways that won't work.”
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Check out voip.ms and see if you can have CF-Busy and CF-No Answer to different locations. Then you set the destination of line 1 to line 2 and line 2 to line 3 on busy and VM for the destination of all 3 lines on no answer.
Sometimes, they try to save you and not allow the 2nd hop, so you need to test to make sure that you can ring 1 then 2 then 3. It may die after the first forward :-)
Carl
This model is end of life
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I assume you don't have your own AA and voice messaging system off your KX-TD 1232? Relying on your service provider's voice messaging for multiple lines like this is an exercise in futility. It is not designed for it and from what you describe, it never worked for you as a proper business voice messaging.
The idea is that a messaging system should answer any ringing line with your company greeting then give the caller a list of extensions from which to choose. When the caller dials that extension it rings the phone on that persons desk and if it isn't answered goes to that person's mailbox. Other options can also be programmed.
Does that sound like the nonsense you have?
-Hal
CALIFORNIA PROPOSITION 65 WARNING: Some comments made by me are known to the State of California to cause irreversible brain damage and serious mental disorders leading to confinement.
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Thanks so much for all the feedback. I have had this issue with other VoIP providers with hunting. Its as if they just don't understand what I am talking about.
OBT: I understand we are using VoIP with an analog system. We have ATA's for each of the lines that plugin to the analog system. I wish they made a expansion unit which supported SIP on this system but I think its just too old? The reason why we put this system in is because we were told they are rock solid and we have phones in 20 classes for communication with the office so to replace all those handsets and the pbx would probably be out of our budget for now. If we did this I would probably just get SIP phones and have hosted phones. We also use our phone system as an intercom system which for emergencies and I don't want to have to rely on the internet for the emergency broadcast PA to work. I know we could look at Asterisk but I am trying to avoid that.
Carl: The last provider I tried to set this up with (callcentric voip provider) basically said the only way to call hunt in the way you are describing is through an unsupported method, they went on to show me how to setup the primary phone line to ring and if busy it would forward the call to an additional line and if it was not answered voicemail would pick up. So instead of a true call hunt roll over it was basically call forwarding if busy on the first three lines and the last was voicemail if busy and voicemail if no answer. This does not seem ideal either. From my experience with ANALOG PBX's most of them want call hunt for this type of scenario.
hbiss: we are not currently using an auto attendant because the office wants a live person to pick up the phone any time it rings so just 8 office phones ring and the other phones in the classes do not.
The other issue I am having is with outgoing calls. I want to setup each extension to essentially be bound to that phone line (ATA FXO technically). So outgoing calls, I would have to have a separate TRG for each extension with 1 CO on each I guess? For example EXT101 would be on TRG1 which would have CO1 in it. The issue this presents is that then people can not do conference calling and such I guess? That seems silly.
I could potentially replace just the Office phones with SIP phones and then somehow connect into the existing PBX for all the classroom phones but I can't really think of a good way to do that. The classroom phones could just dial out and it would ring all the extensions in the office but for the office to call the intercom function or classroom phones, I have no idea how that would work. Is there a way to dial into the PBX from an outside line and tap into the intercom function? Oh one idea is to use one of the analog Jacks to tap into the intercom system. Use an ATA connected to an analog jack and from the SIP office phones they could reach the ATA and then dial 33* to page the whole school? At least this way we could budget and work towards replacing equipment in phases (first the office phones, then later all the classroom phones)
-Jemmyn [Panasonic KX-TD1232]
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hbiss: we are not currently using an auto attendant because the office wants a live person to pick up the phone any time it rings so just 8 office phones ring and the other phones in the classes do not. You are not well informed. With a messaging system you don't have to use the AA function if you don't want to. Calls on any line can be answered manually as above but you would have calls that aren't answered in, say, 4 rings hear a menu or just be forwarded to a mailbox. NOT a mailbox for every line! You want the DISABLE your service provider's stupid ring-no answer forwarding or, if that can't be done at least make it transfer at the maximum number of rings- usually 10. That way your messaging system will pick up after 4 (or more) rings in plenty of time. Here, spend the 110 bucks and do yourself a favor. https://panasonicbts.factoryoutlets...ategory_id=19376&catalogitemid=60751The other issue I am having is with outgoing calls. I want to setup each extension to essentially be bound to that phone line (ATA FXO technically). So outgoing calls, I would have to have a separate TRG for each extension with 1 CO on each I guess? For example EXT101 would be on TRG1 which would have CO1 in it. The issue this presents is that then people can not do conference calling and such I guess? That seems silly. Someone that knows how to program the TD1232 will have to talk about this. I know with other systems I can have lines assigned to extensions no problem. I think your problem is that you are using pools. -Hal
CALIFORNIA PROPOSITION 65 WARNING: Some comments made by me are known to the State of California to cause irreversible brain damage and serious mental disorders leading to confinement.
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First of all, wind this conversation back to the "hunting" part. I use Grandstreams. Like https://www.callcentric.com/support/device/grandstream/ht814 or the HT818. They are set to hunt by programming and you get a pilot number with x amount of ports. In my service provider, (Ipitomy or Flowroute) they don't care how many ports I have as I pay by the minute for service, or buy some block of minutes. I can have 3, or 10 or 20 calling paths if I have the capacity to handle that many. As Hal pointed out, a TVS-XX voice mail box works...if you have the right version TD system, it integrates pretty smoothly. I have 3 or 4 of them on the shelf at a price way south of the $110 street price. Even Ebay is littered with TVS boxes for not much money. That gives you the voice mail part and lights the message waiting lamp of the receiver or master phone. Oh yeah, I saved the TD-1232's, fully carded and at least v5. Tick tock, before they go on Ebay :-) Anyway, take a close look at the provisioning of the SPA-8000 box. I think it should program similarly to the HT-818. Carl
This model is end of life
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hbiss: Thanks for replying. I understand that you don't have to have Auto Attendant features enabled to utilize voicemail boxes. We have just never used more then one voicemail box and we have really enjoyed features like emailing the .wav of the voicemail and transcribing which of course you cant get with the TVS50. We actually have the TVS50 and it was actually setup with our system at one point and the internal hard drive failed and I was never able to get it set back up. I remember it being a bit of a PITA to setup though. I would love to get it setup correctly again.
Carl: This is a really helpful tip. Perhaps I need to send this Cisco back and get a Grandstream. I consider one when purchasing this. The big question is... How do you have the service provider setup? Does it just ring all the FXO ports at the same time and then the Grandstream takes care of the proper call hunting. Do you also use the a local voicemail solution instead of hosted on in this scenario? At our SIP provider we currently have the main account and then 8 sub accounts and each sub account is hooked up to one of the FXO ports, they all use the main lines minutes by they are able to be configured slightly differently for each line. Should I be doing it this way or should each FXO be set to use the same main account DID credentials?
Last edited by jemmyn; 04/17/19 10:28 AM.
-Jemmyn [Panasonic KX-TD1232]
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I think I was able to answer my own question. At first the Linksys / Cisco SP8000 did not have any trunking options so I had to go to cisco's website and follow the tutorial on updating the firmware on the unit. https://www.cisco.com/c/en/us/suppo...ttings-on-the-spa8000-phone-adapter.htmlAfter I did that the trunking options showed up. * I went to Sipura web interface by typing the IP of my unit into Chrome, clicked on Admin login and Advanced Clicked on the SIP tab scrolled down to bottom to ensure Trunk Parameters section Hunt Policy was "onhook only" * Then I went to T1 tab at the top (Trunk 1) * It essentially gets setup like a single SIP DID Line (FXO ports) would be so it has all the pertinent parameters that would need to setup with your SIP provider like proxy, port, User ID, Password, etc. I mirrored my settings of my main DID * It also had a section called Contact List: which was set to 1,2,3,4,5,6,7,8,hunt=re;*;1 (appearantly to hunt through all the lines) * I submitted changes and it took a lot longer to refresh then the 5 seconds it claims it needs. Then when it got back up all my lines could dial out (over the main DID) and now when I call in it starts on line 1 if its busy line 2 if its busy line 3 and so on. I verified that my main DID can handle up to 25 concurrent calls. This is something that I assume could potentially throw off this functionality. My provider VoIP.ms has an unlimited subscription DID which only supports 3 concurrent calls which I am sure would not work if you wanted all 8 lines in the Trunk (Hunt Group) as I do. Perhaps you could have 3 in the Trunk in that case. I imagine the Grandstream interface is much less combersom and more user friendly but I think its working. I also had to go into the Panasonic software (KX-TD Programmator) and configure the TRG #1 to include all of the lines and then I went into each extension for the office and set lines 1-8 to ring immediately on Day and Night. I have not been able to get my Dial Plan working that was setup on the individual Lines/FXO ports so I just left that as at default ([*#0-9A-D][*#0-9A-D].) which means I have to dial the whole number with the 1 on the front. Anyway thanks for all your feedback, it is nice to know that they have added this functionality now to my ATA and that I don't have to rely on my SIP provider to have the option. I really appreciate this forum. Every time I use it you guys do an amazing job of pointing me in the right direction. Now on to that KX-TVS50...
Last edited by jemmyn; 04/17/19 12:07 PM. Reason: formatting
-Jemmyn [Panasonic KX-TD1232]
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Just a quick update to let you guys know that the Linksys / Cisco SPA 8000 Call hunt feature works perfectly after the firmware update. We have been testing it with VoIP.ms for the last week or so without any issues and it is fantastic in combination with our Panasonic KX - TD 1232. The only thing I would caution is about the VoIP support staff. They have no idea what call hunt is or any other ways to implement it, they even cautioned me over and over not to setup my SPA8000 to login call hunt for all the lines, then when I told them I had it setup this way for a week without any issue they said "Oh well then disregard my (vigorous) cautions." Thanks for all the help guys and for your recommendation Carl. Good stuff. Let me know if I can do anything to give back. tikimultimedia.com is my company.
Last edited by jemmyn; 04/29/19 02:38 PM.
-Jemmyn [Panasonic KX-TD1232]
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