18 Replies Latest reply on Jul 21, 2016 10:48 AM by markfreeman

    Cordless phone

    buffit New Member

      I need to get a reasonably priced cordless phone that will work with my UCC server and Adtran system.

       

      Any suggestions?

        • Re: Cordless phone
          jayh Hall_of_Fame

          SIP based or analog to FXS port?  Wi-fi, DECT, or other?  Lots of choices.

           

          Good, fast, cheap.  Choose two.

            • Re: Cordless phone
              buffit New Member

              Sip,  I was looking at the Aastra_Model_480i_CT, but all I can find are used ones.


              The other question is do I need both the full phone and then the cordless? or can I just get the cordless part?

              don't want cheap cheap, but reasonable, $150 or so.


              IT is going into my Adtran VOIP system, so I assume SIP. 



                • Re: Cordless phone
                  jayh Hall_of_Fame

                  We've had good results with the Panasonic KX-TGP500.

                   

                  There are three parts to it.

                  • RF unit - Connects to Ethernet LAN and contains the base radio transmitter/receiver.  This should be located to cover the desired area where the handset(s) will roam.
                  • Handset - The actual phone carried by the user
                  • Charging cradle - Stand that holds the handset when not in use and charges the battery.

                   

                  A spare handset and charger pair is available as part number KX-TPA50.  A single RF unit will support multiple handsets. 

                   

                  These don't support LLDP or CDP so if you use a separate voice VLAN (and you typically should), then you'll need to assign a switch access port statically to the voice VLAN for it. 

                   

                  "Cordless phone" to most vendors implies a consumer-type unit typically for analog lines.  Many of the cheaper ones have the RF unit built in to the same enclosure as the charger which can be a problem if the optimum location for radio signals isn't convenient for charging the phone.

                  1 of 1 people found this helpful
              • Re: Cordless phone
                buffit New Member

                ok I ordered the phone and now have a setup question.

                 

                Just an FYI the manual/quick guide that comes with the Panasonic is horrible.

                 

                So, I have the KX-TGP500 (the base station with 1 phone)

                The phone has gotten an ip address from the network which is good.

                But, when I goto set up the phone, it can not find what type of phone it is.  I tried doing it as unknown and calling it and it didn't work.  So, how do I tell my UCC server what type of phone?  There was no CD that I can find in the box.

                 

                Just neeed some help with general setup for this phone.  I know how to create the user and all that.

                  • Re: Cordless phone
                    Employee

                    jayh may have more input to add since he has used this particular phone before.  In general if it is a phone not supported by NetVanta UC, you can create a new user and just skip the step to assign a phone.  Then you would have to provision that phone manually or through some method other than UC.  The SIP authentication username and password configured on the phone need to match the UC identity.

                     

                    Thanks,

                    Matt

                    • Re: Cordless phone
                      jayh Hall_of_Fame

                      It probably won't auto-provision unless there is support for it in the 7000 which it sounds like there isn't.  You'll need to load its SIP parameters manually or from a separate TFTP server.

                        • Re: Cordless phone
                          buffit New Member

                          I am logged into the handset and have the web gui up

                          here are the current settings

                           

                          Not sure what the settings are I am missing.

                          VOIP

                          SIP Settings

                          Line 1

                          Phone number 1374

                          Line ID 1374

                          Registrar Server Address
                          Registrar Server Port [1-65535]
                          Proxy Server Address
                          Proxy Server Port [1-65535]
                          Presence Server Address
                          Presence Server Port [1-65535]

                          Outbound Proxy Server

                          Outbound Proxy Server Address
                          Outbound Proxy Server Port [1-65535]
                          SIP Service Domain
                          Service Domain
                          SIP Source Port
                          Source Port [1024-49151]
                          SIP Authentication
                          Authentication ID
                          Authentication Password
                          DNS
                          Enable DNS SRV lookupYesNo
                          SRV lookup Prefix for UDP
                          SRV lookup Prefix for TCP
                          Timer Settings
                          T1 Timer milliseconds
                          T2 Timer seconds
                          INVITE Retry Count
                          Non-INVITE Retry Count
                          Quality of Service (QoS)
                          SIP Packet QoS (DSCP)
                          SIP extensions
                          Supports 100rel (RFC 3262)YesNo
                          Supports Session Timer (RFC 4028) seconds [60-65535, 0: Disable]
                          Keep Alive
                          Keep Alive Interval seconds [10-300, 0: Disable]
                          Security
                          Enable SSAF (SIP Source Address Filter)YesNo
                            • Re: Cordless phone
                              jayh Hall_of_Fame

                              buffit wrote:

                               

                              I am logged into the handset and have the web gui up

                              here are the current settings

                               

                              Not sure what the settings are I am missing.

                              VOIP

                              SIP Settings

                              Line 1

                              Phone number 1374

                              Line ID 1374

                              Registrar Server Address
                              Registrar Server Port [1-65535]
                              Proxy Server Address
                              Proxy Server Port [1-65535]
                              Presence Server Address
                              Presence Server Port [1-65535]

                              Outbound Proxy Server

                              Outbound Proxy Server Address
                              Outbound Proxy Server Port [1-65535]
                              SIP Service Domain
                              Service Domain
                              SIP Source Port
                              Source Port [1024-49151]
                              SIP Authentication
                              Authentication ID
                              Authentication Password
                              DNS
                              Enable DNS SRV lookup YesNo
                              SRV lookup Prefix for UDP
                              SRV lookup Prefix for TCP
                              Timer Settings
                              T1 Timer milliseconds
                              T2 Timer seconds
                              INVITE Retry Count
                              Non-INVITE Retry Count
                              Quality of Service (QoS)
                              SIP Packet QoS (DSCP)
                              SIP extensions
                              Supports 100rel (RFC 3262) YesNo
                              Supports Session Timer (RFC 4028) seconds [60-65535, 0: Disable]
                              Keep Alive
                              Keep Alive Interval seconds [10-300, 0: Disable]
                              Security
                              Enable SSAF (SIP Source Address Filter) YesNo

                              Well I tried to edit your form but nothing stuck...

                               

                              Line-ID should match the SIP identity of the device on the 7000

                              Registrar and all proxy server addresses should be the IP of the 7000 inside

                              Set all ports to 5060

                              No DNS SRV if behind a single non-redundant system.

                              Domain is your SIP domain/realm.

                              Authentication ID should match the 7000 sip auth-name

                              Authentication password should match the 7000 sip password

                               

                               

                              That should get you going if it's inside the NAT of your 7000 series.  Couldn't edit some fields but SSAF would likely be good to check but shouldn't matter if behind a NAT.  SIP source port can be any high port, 5060 will be fine.  Line ID should match the SIP identity of the station, typically the phone number.

                        • Re: Cordless phone
                          buffit New Member

                          Sorry for the late reply...thank you for all your help!!!

                          Apparently the one step that was left out...which I didn't know till I accidentally did it.  After you make all the changes to the cordless handset and to the system.  you have to reboot the base station.

                           

                          Thanks again for the help.

                            • Re: Cordless phone
                              grimard New Member

                              I have successfully connected a KX-TGP500 to our NetVanta 7100, however I have an issue, calls drop after 60 seconds. Any thoughts ?

                                • Re: Cordless phone
                                  markfreeman Employee

                                  Log into the CLI of the 7100 and issue the following debug commands and then place a call and stay online till the call drops and them post the debug. Make sure to note phone numbers that are being used.

                                   

                                  Debug sip stack messages

                                  Debug voice verbose

                                  Debug sip cldu

                                   

                                  -Mark

                                    • Re: Cordless phone
                                      grimard New Member

                                      Here is what I get when extension 625 (KX-TGP500) calls extension 231 (Adtran softphone) and the call is dropped after exactly 1 minute:

                                       

                                      13:16:51.513 PM.625 Ca:0 Sending Keep-alive: INFO

                                      13:16:51.516 PM.625 Ca:0 call-leg transaction -> Request Sent

                                      13:16:51.522 PM.231 Ca:0 Sending Keep-alive: INFO

                                      13:16:51.525 PM.231 Ca:0 call-leg transaction -> Request Sent

                                      13:16:51.595 PM.625 Ca:0 SipPM_Connected      rcvd SIP call-leg response: 501 Not Implemented

                                      13:16:51.596 PM.625 Ca:0 call-leg transaction -> Final Response Rcvd

                                      13:16:51.596 PM.625 Ca:0 SipPM_Connected      ERROR! Received 501 error for keep-alive, clearing call.

                                      13:16:51.596 PM.625 Ca:0 State change      >> SipPM_Connected->SipPM_Terminating

                                      13:16:51.596 PM.625 Ca:0 SipPM_Terminating    sent: SA->Appearance Off

                                      13:16:51.598 PM.625 Ca:0 SipPM_Terminating    call-leg (P:0x5b73928 S:0x0) -> Disconnecting (Local Disconnecting)

                                      13:16:51.598 PM.625 Ca:0 SipPM_Terminating    sent: BYE

                                      13:16:51.598 PM.625 Ca:0 call-leg transaction -> Terminated

                                      13:16:51.598 PM.625 Ca:0 SipPM_Terminating    ERROR! SipCallLegTransactionStateChanged to Terminated ignored

                                      13:16:51.599 SA.625 Ca:0 Connected            rcvd: AcctPhoneMgr_appearance(OFF) from PM

                                      13:16:51.599 SA.625 Ca:0 Connected            sent: clearCall to SB

                                      13:16:51.599 SA.625 Ca:0 Connected            State change      >> Connected->Clearing (CAS_Active)

                                      13:16:51.599 SB.CALL 5043 Connected            Called the clearCall routine

                                      13:16:51.600 SB.CALL 5043 Connected            ClearCall sent from 625 to 231

                                      13:16:51.600 SB.CALL 5043 State change      >> Connected->Clearing

                                      13:16:51.600 SA.231 Ca:0 Connected            rcvd: clearCall from SB

                                      13:16:51.600 SA.231 Ca:0 Connected            sent: clearResponse(pass) to SB

                                      13:16:51.601 SA.231 Ca:0 Connected            State change      >> Connected->Idle (CAS_Idle)

                                      13:16:51.601 SA.231 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                                      13:16:51.601 PM.231 Ca:0 State change      >> SipPM_Connected->SipPM_Byeing

                                      13:16:51.603 PM.231 Ca:0 SipPM_Byeing         call-leg (P:0x5b73088 S:0x0) -> Disconnecting (Local Disconnecting)

                                      13:16:51.603 PM.231 Ca:0 SipPM_Byeing         sent: BYE

                                      13:16:51.604 PM.231 Ca:0 State change      >> SipPM_Byeing->SipPM_Terminated

                                      13:16:51.604 PM.231 Ca:0 State change      >> SipPM_Terminated->SipPM_Idle

                                      13:16:51.604 SB.CALL 5043 Clearing             Called the clearResponse routine

                                      13:16:51.604 SB.CALL 5043 State change      >> Clearing->CallIdlePending

                                      13:16:51.605 SB.CCM disconnect:

                                      13:16:51.605 SB.CCM  :  Call Struct 0x6d5d810 :   Call-ID = 5043

                                      13:16:51.605 SB.CCM  :  Org Acct = 625    Dst Acct = 231

                                      13:16:51.605 SB.CCM  :  Org Port ID = SipPhone 0/0   Dst Port ID = SipPhone 0/0

                                      13:16:51.606 MOH.APP printCSHoldStates, disconnect: towOrig 0, towDest 0 origHold 0 destHold 0 isHold 0

                                      13:16:51.606 SB.CCM disconnect: Call Connection Type is RTP_TO_RTP

                                      13:16:51.606 SB.CCM release:

                                      13:16:51.606 SB.CCM  :  Call Struct 0x6d5d810 :   Call-ID = 5043

                                      13:16:51.606 SB.CCM  :  Org Acct = 625    Dst Acct = 231

                                      13:16:51.607 SB.CCM  :  Org Port ID = SipPhone 0/0   Dst Port ID = SipPhone 0/0

                                      13:16:51.607 SB.CCM release: Call Connection Type is RTP_TO_RTP

                                      13:16:51.607 SB.CALL 5043 CallIdlePending      ClearResponse sent from 231 to 625

                                      13:16:51.607 SA.625 Ca:0 Clearing             rcvd: clearResponse from SB

                                      13:16:51.608 SA.625 Ca:0 Clearing             State change      >> Clearing->Idle (CAS_Idle)

                                      13:16:51.608 SA.625 Ca:0 Idle                 sent: AcctPhoneMgr_cachg(CAS_Idle) to PM

                                      13:16:51 SB.CallStructObserver 5043 Finalized

                                      13:16:51.698 PM.625 Ca:0 SipPM_Terminating    rcvd SIP call-leg response: 200 OK

                                      13:16:51.699 PM.625 Ca:0 SipPM_Terminating    call-leg (P:0x5b73928 S:0x0) -> Disconnected (Disconnected)

                                      13:16:51.699 PM.625 Ca:0 State change      >> SipPM_Terminating->SipPM_Terminated

                                      13:16:51.699 PM.625 Ca:0 State change      >> SipPM_Terminated->SipPM_Idle

                                      2016.07.21 13:16:52 SMDR 5043       07/21/2016 13:15:48      1.0 0    I  00/00 Jacques Grimard 625             00/00 Pie

                                      rrot Robert  231             0 N

                                        • Re: Cordless phone
                                          markfreeman Employee

                                          One thing you could try not sure who user you got the 501 message from but you could try turning off keepalive, by default we send those every 60 seconds, maybe your cordless phone doesn’t like it. Just copy and paste the two commands below into global configuration mode. Make sure you save if it works.

                                           

                                          Voice user 625

                                          No sip-keep-alive

                                           

                                          If that doesn’t work then I will need the full debug, you just send me the end, please run this and post all of it.

                                           

                                          Debug sip stack messages

                                          Debug voice verbose

                                          Debug sip cldu

                                           

                                           

                                          -Mark