Cordless phone

[buffit]

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

Any suggestions?

[jayh]

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

Good, fast, cheap.  Choose two.

[buffit]

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. 

[jayh]

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.

[buffit]

OK, so from your post I would need something like

Panasonic KXTGP500B04 DECT 6.0 Cordeless Phone

That is the RF unit that plugs into the network, and 1 cordless handset?

[jayh · Accepted solution]

buffit wrote:

OK, so from your post I would need something like

Panasonic KXTGP500B04 DECT 6.0 Cordeless Phone

That is the RF unit that plugs into the network, and 1 cordless handset?

Yes, as well as one charging cradle.  See previous post if you need more than one handset.

[buffit]

Thank you for the help....it is on order

[buffit]

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.

[Anonymous]

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

[jayh]

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.

[buffit]

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

[jayh]

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.

[buffit]

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.

[grimard]

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

[Anonymous]

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

[grimard]

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

[Anonymous]

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

[grimard]

Hi,

You must be some kind of genius, "No sip-keep-alive" solved the issue. Thank you very much.

[Anonymous]

Good to hear! Thanks for responding back and letting us know that fixed your problem.

-Mark