atcomsystems.ca/forum Forums Active Threads

Re: NEC announcement R4+Z Yesterday at 12:24 PM

We have heard the same thing in OZ!

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Re: Partner Endeavor Andyreed Yesterday at 12:07 AM

Didn't the Endeavor phones have tiny little displays on them?

Seems like the Partner was plenty economical as is....

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Other Telephones & Systems

Re: IPO Ip phone install Andyreed Yesterday at 12:05 AM

An H.323 phone. Pretty straightforward, not really any different than adding a digital phone. Add a new user in Manager, and if the extension you use for the user does not exist it will ask you if you want to create it.

From there, it really all depends on how the customer's network is setup. If they already have IP phones there's a good chance all you will need to do is plug it in to a powered ethernet port, wait for it to do a firmware upgrade, and then login with the extension/password you created.

If it's the first IP phone that's a whole other ball of wax.

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Cancel forwarding on ISDN 8510T Keyset6 04/18/24 06:38 PM

I realize this is 40+ year old technology. I have a question into my telco contact that's familiar with the AT&T 5ESS, which is somewhat different than ours. My customer's remote office has a few AT&T ISDN 8510T phones served from the local CO. The customer wants to forward their main number at times to another phone. The main number is not the primary line on an ISDN phone but is a SLO (Single Line Only) appearance on 2 phones. So, the typical procedure of 72#, (destination), wait for confirmation tone, press the CFV key does not apply.

She successfully forwarded the number by 174+(XXXX, the 4 digit extension). But, the code to cancel forwarding is unknown. In our system, an SLO is forwarded by 78#+ the 5 digit extension, 79# to cancel. The remote office gets a fast busy immediately when pressing 7, and 179+ 5 digits did not work. The CO serving the remote customer is a public CO, and I don't know if it's the same vintage as ours. My contact hasn't been able to research the SLO call forward cancel procedure yet, so I thought I'd give it a shot here.

Re: Looking for TIE KSU Andyreed 04/18/24 06:37 PM

Well, so far I’ve gotten close. Kevin, C5Z, knows the whereabouts of a TCX-128 system. I guess you could call it the evolution of the 2260. He’s going to help coordinate that for me and I greatly appreciate it.

I found the power supply for the 2260 but no KSU so far. I’ve also found a TC-12 KSU (as far as I can tell it’s essentially a 1236 but uses 2 pairs?), but no power supply for it. It looks like it may be compatible with the 2260’s power supply but can’t tell for sure. Power specs in the manual aren’t detailed enough for me to tell.

So, still looking for an 820/1236/2260 if anyone runs across one. And in the meantime I look forward to getting the TCX-128 running.

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Re: Revive a OS7070 Toner 04/17/24 07:42 PM

VERY old thread - locking.

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Re: OfficeServ 7030 stuck on boot Toner 04/17/24 07:41 PM

Old thread - it is now locked.

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Re: IPO Manager Toner 04/17/24 06:47 PM

I've uploaded it to the Avaya Manuals section: https://atcomsystems.ca/forum/ubbth...ya-ip-office-software-r8-1-95#Post646240

You should be able to do a 'trial' subscription for $1 and cancel within the first 7 days: https://sundance-communications.com/forum/ubbthreads.php/ubb/subscription/id/16

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General Telephone & Systems Information

Re: SV9100 and Baum's Act programming newtecky 04/16/24 09:24 PM

Originally Posted by teldata1

I thought Baums law was to get the location i.e. room number
Which would have to be done with a E911 Sip Trunk from what I understand

Lots of programming on both ends, the NEC and the provider

A lot of the hosted companies are throwing this into the mix and people don’t think twice. They just go with the hosted product if they have any money.

Seems like a ridiculous law to me. Everyone has a cell phone the NEC can tell you where the call was made.

Obviously, it would apply in certain circumstances

RAY BAUM’s Act requires that all 911 calls must have a “dispatchable location” which means having information for emergency responders to find the exact location of a person who has dialed 911. In a standard office with a reception, I don't think there is much that needs to be done, expecially if the PBX notifies the reception about who dialed 911. In larger locations or some specific circumstances (as you mentioned) it would make sense to use multiple e911 numbers for specific location information.

We had 2 offices that shared the same PRI. We worked with the carrier to send 2 different e911 numbers with the correct address depending on which building the extensions were located.

Our company is doing nurse care facilites with phones in every rooms. With a hosted system we are now able to provide a DID to every internal extensions, and register every phone number with with a room number along with the physical address for 911. The nurses also get notified if someone dials 911, just like most on premisse PBXs already do as well.

The main issue with cell phones is that there is no location information and I don't beleive that they are routed to a local 911 dispatch center. In a panic, people may not be able to rememeber the address of their current address. Also if the call gets cut off before they can tell the address to the dispatcher.

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Re: Oddball modular connectors RJ45? DonaldR 04/11/24 11:52 PM

If the information you found is from the typewriterdatabase, that is the information I recently shared.

I thought it might be a standard alternate type connector rather than a "proprietary designed" which IBM seems to love so much.

I can see why it would require the key. You wouldn't want someone plugging a 10 Base-T modular connector into the back of the IBM 85/95 Electronic Typewriter, or the 6731. Who knows what might happen.... Bzzzzzz *POOF!*

Things I didn't know about GRANDSTREAM!! Carl Navarro 04/06/24 01:16 AM

I picked up some Grandstream material in the last week and my surprise began when I couldn't register the secondary stuff to my Cloud account. It appears that they addressed this issue last year. https://www.grandstream.com/online-marketplace-warranty-limitations-policy-page

Oops. I suppose it's fair, but GS doesn't want any of the product that you can administer in the cloud to be purchased or resold on the secondary market.

Meanwhile, I have a couple of gray systems and 10 or so phones of questionable heritage. :-( Time to figure out a workaround...

Carl

EDIT: After wailing and gnashing of teeth and texting Grandstream, I am now able to log into my cloud account and add this unit to my systems. The seller (Phillips Communications) on Ebay contacted who he bought it from and they released my new system back into the pool. Now it's mine. It's the first time they have had to deal with the 6300 series systems.

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Re: SV9100 CCIS Issues erniejoey 04/05/24 07:31 PM

FYI...The network engineers ran Wireshark with call examples. The network engineers had me reboot both SV9100's and after that it works fine now. The network engineers said that when they installed their new equipment sometimes the MAC Address's do not release from the NEC equipment. Never seen that before.

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IPO R11.1.3.1.0 DTMF Issue with J1xx phones Toner 04/05/24 04:51 PM

Hi all,

FYI and for our own reference we've found what appears to be a very specific DTMF bug with R11.1.3 and J1xx phones (it doesn't affect 9608 phones). Here's the rundown:

A call that is placed which has an immediate shortcode match results in the IPO telling the J1xx phones to use RFC2833 for DTMF. In this example the phone uses it's redial button to place a call:

Code

2024-04-04T17:41:01   56533469mS CMExtnEvt:    v=0 State, new=Dialling  old=Idle,0,0,80102 AtcomJ1xx
2024-04-04T17:41:01   56533469mS CMTARGET:     0a0597eb30024522 866.805455138.0 17731 80102 AtcomJ1xx.0: LOOKUP CALL ROUTE: GID=0 type=100 called_party=9055551234 sub= calling=8010229 calling_sub= dir=out complete=1 ses=0
2024-04-04T17:41:01   56533469mS CMTARGET:     0a0597eb30024522 866.805455138.0 17731 80102 AtcomJ1xx.0: ADD TARGET (N): number=9058299444 type=100 depth=1 nobar=1 setorig=1 ses=0
2024-04-04T17:41:01   56533470mS CMTARGET:     0a0597eb30024522 866.805455138.0 17731 80102 AtcomJ1xx.0: DEF SC: 9055551234 0  sc=type=Dial code=?, num=.s15875551234
2024-04-04T17:41:01   56533470mS CMTARGET:     0a0597eb30024522 866.805455138.0 17731 80102 AtcomJ1xx.0: INITIAL TARGETING SUCCEEDED
2024-04-04T17:41:01   56533470mS CMTARGET:     0a0597eb30024522 866.805455138.0 17731 80102 AtcomJ1xx.0: GetNoAnswerTimer:20
2024-04-04T17:41:01   56533470mS CMExtnEvt:    v=0 State, new=Proceeding  old=Dialling,0,0,80102 AtcomJ1xx

First, the IPO placed the call over the SIP trunk. It then sends an INVITE to the J1xx phone (I know, this seems backwards but it's how CCMS signaling seems to work)

Code

2024-04-04T17:41:01   56533757mS SIP Tx: TLS 10.0.0.1:5061 -> 10.0.0.2:5429
                    INVITE sip:[email protected]:5429;transport=tls SIP/2.0
                    v: SIP/2.0/TLS 10.0.0.1:5061;rport;branch=z9hG4bKe81d753a9840e4a9997cda7d166df659
                    f: <sip:[email protected]>;tag=5a4531af994bfd80
                    t: <sip:[email protected]>;tag=660e61bc532561211k6s6l6u1g3kd4n1n415573_F8010229
                    i: 1_660e61bc-4833d3c75u26o3h2h56q331zi4w1u_I8010229
                    CSeq: 574 INVITE
                    m: <sip:[email protected]:5061;transport=tls>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
                    User-Agent: IP Office 11.1.3.1.0 build 34
                    c: application/sdp
                    l: 203
                    
                    v=0
                    o=UserA 1563834215 396372004 IN IP4 10.0.0.1
                    s=Session SDP
                    c=IN IP4 10.0.0.1
                    t=0 0
                    m=audio 50700 RTP/AVP 0 101
                    a=rtpmap:0 PCMU/8000
                    a=rtpmap:101 telephone-event/8000
                    a=fmtp:101 0-15

You can see that RFC2833 is specified with the line "a=rtpmap:101 telephone-event/8000"

Now a call to a 10 digit number where there is no specific shortcode match and we have to wait for the 4 second timeout:

Code

2024-04-04T17:40:40   56511104mS CMExtnEvt:    v=(null) State, new=Idle  old=Idle,0,0,8010229: Digit Key Pressed 4
2024-04-04T17:40:40   56511105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: Setting Hard Timer 4000
2024-04-04T17:40:40   56511105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: LOOKUP CALL ROUTE: GID=0 type=100 called_party=9055551234 sub= calling=8010229 calling_sub= dir=out complete=0 ses=0
2024-04-04T17:40:40   56511105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: ADD TARGET (N): number=90555512344 type=100 depth=1 nobar=1 setorig=1 ses=0

(4 seconds later)
2024-04-04T17:40:43   56515105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: TimerExpired cause=CMTCDelayedProcessing
2024-04-04T17:40:43   56515105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: LOOKUP CALL ROUTE: GID=0 type=100 called_party=9055551234 sub= calling=8010229 calling_sub= dir=out complete=1 ses=0
2024-04-04T17:40:43   56515105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: ADD TARGET (N): number=9055551234 type=100 depth=1 nobar=1 setorig=1 ses=0
2024-04-04T17:40:43   56515105mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: DEF SC: 9055551234 0  sc=type=Dial code=?, num=.s15875551234
2024-04-04T17:40:43   56515106mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: INITIAL TARGETING SUCCEEDED
2024-04-04T17:40:43   56515106mS CMTARGET:     0a0597eb30024509 866.805455113.0 17724 80102 AtcomJ1xx.0: GetNoAnswerTimer:20
2024-04-04T17:40:43   56515106mS CMExtnEvt:    v=0 State, new=Proceeding  old=Dialling,0,0,80102 AtcomJ1xx

This time the IPO starts with an INVITE to the J1xx phone but is missing the rFC2833 line and therefore we would expect Inband DTMF.

Code

2024-04-04T17:40:43   56515106mS SIP Tx: TLS 10.0.0.1:5061 -> 10.0.0.2:5429
                    INVITE sip:[email protected]:5429;transport=tls SIP/2.0
                    v: SIP/2.0/TLS 10.0.0.1:5061;rport;branch=z9hG4bK4f167ffffdfbb2dd55f6bc0679c87cea
                    f: <sip:[email protected]>;tag=5a4531af994bfd80
                    t: <sip:[email protected]>;tag=660e61bc532561211k6s6l6u1g3kd4n1n415573_F8010229
                    i: 1_660e61bc-4833d3c75u26o3h2h56q331zi4w1u_I8010229
                    CSeq: 562 INVITE
                    m: <sip:[email protected]:5061;transport=tls>
                    Max-Forwards: 70
                    Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,SUBSCRIBE,REGISTER,PUBLISH,UPDATE
                    User-Agent: IP Office 11.1.3.1.0 build 34
                    c: application/sdp
                    l: 121
                    
                    v=0
                    o=UserA 1563834215 396372001 IN IP4 10.0.0.1
                    s=Session SDP
                    c=IN IP4 10.0.0.1
                    t=0 0
                    m=audio 0 RTP/AVP 0

The IPO then places the call over the SIP trunk and specifies RFC2833.

Shortly afterward, the J1xx responds with a 200 OK that does not contain the RFC2833 line.

A capture of the RTP packets from the J1xx to the IPO proves the DTMF is missing in both methods - RFC2833 and Inband.

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Re: Upgrade Samsung OS 7030 to ver 5.03 pvj 04/04/24 03:31 AM

Do you have the 4.93 upgrade procedure document? You can use file control to upload the 5.03 file to the system.

A quick setup , really not.Every system is programmed depending on how a customer wants it to work. You can default the system to start from scratch.

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Re: Where would I find eManager for a 670? Toner 04/02/24 04:37 PM

Private message sent.

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Request a Telephone Technician

Programming OfficeServ 7xxx to forward incoming calls highlysecptial 03/31/24 02:37 PM

Sorry if this is a very naive question, but can anyone explain to me how to program the OfficeServ Device Manager (I have an OfficeServ 7100) to forward all incoming calls on specific incoming trunks to another external number? Ideally, I would also like to specify which trunk is used for the diverted calls.

Any help would be gratefully received.

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Antique Telephones

Re: Payphone Assistance BPopilek 03/30/24 02:09 AM

Thank you for the suggestion. I was finally able to give that a try this evening. Unfortunately, I still get nothing from the handset. Is there a way to test the hand piece and the buttons? I'm assuming something using a multimeter with continuity tester? If I can troubleshoot my way through, hopefully I can find a way to get it working again.

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Re: Old guy (Mitel installation) near Dayton, OH hitechcomm 03/28/24 11:23 PM

VERY sMART

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General Telephone & Systems Information

Re: Fluke Netscout LRAT-2000 Repair Service?? DuddlyDoRight 03/27/24 02:04 PM

I did give them a try, unfortunately they aren't taking on repairs of units at this time, just looking to sell I guess. Thanks for the input though.

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Re: NEC DS1000 Rings to Answer dans 03/26/24 12:15 PM

I apologize for the delay in responding back to you nectec and for the incorrect answer, go to the same program area 0401, look for trunk revert, make your change there, by default is set to 15 seconds. Make sure you press the hold key after making the change.

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Re: 5.03 Software Toner 03/22/24 08:21 PM

I'm going to lock this thread. If anyone wants to send me the 5.03 software I'll upload it to the "Manuals" section.

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Re: PAD and CTX 670 jc2it 03/22/24 05:54 PM

I am unsure how it is connected. I know there are four zones and I think the codes #3104, #3105 and #3106 operate some of the zones and there are several others but I don't know them off the top of my head. Each zone is in series. and is comprised of 5 or 6 Valcom V-1030C loudspeaker horns. It was originally ran with CAT3 wire with a pair twisted together for each conductor. When we have troubleshooted it over the years we have replaced it with an 18 ga. shielded and we get better results with that.

What we are experiencing sounds like to much gain on the MICs and it results in feedback in some circumstances and overpowering the loadspeaker in others. If I lower the input on the MIC the loudspeakers are easier to understand. We have done this in two ways.

On Analog phones and DKTs we have put a piece of Scotch tape over the MIC holes in the phone.

Also, we have a Spectralink system connected to the PBX which simulates a DKT2020 and this system allows for noisy environments with three "noise" settings, normal, high and severe. The end result is high, and severe are two different mute settings for the MIC. When I put the Spectralink phone in "High" the sound from the loudspeaker becomes clearer, and "Severe" mode makes it pleasant and almost crystal clear.

It is likely the loudspeaker volume is turned up too high. But because they are in series adjusting the volume on one loudspeaker seems to affect all of them to some varying degree. That is a lot of up and down on a lift, however this may be what we do since testing PAD didn't seem to adjust the volume or muffle the test calls we did. I would probably need a specific procedure to get it to work at this point.

Thanks!

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Re: Dial Number Plan Wizard newtecky 03/22/24 12:07 AM

Win2K? You sure have been babysitting that system for a while.

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