Solved: Interconnection CME with Panasonic KX-TDE 600

chrysostomos1980 · ‎02-10-2012

Hi to all

We have one CME and one Panasonic KX-TDE 600.we need to integrate these PBX via H323 or via SIP protocol

From my site(CME) i have created one dial peer like the below

i think that we this we are ok from our site right?

Do you know anything about Panasonic site?

its asked me A credentials for the sip but we dont use any credential for the sip trunking, dial peer on CME

FOR THE SIP PART

dial-peer voice 19 voip

destination-pattern 1..

session protocol sipv2

session target ipv4:192.168.10.1

dtmf-relay sip-notify

codec g711ulaw

no vad

FOR THE H323 PART

dial-peer voice 21 voip

destination-pattern 1...

session target ipv4:192.168.10.1

dtmf-relay h245-alphanumeric

REGARDS

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""

acampbell · ‎02-10-2012

Hi,

Have a look at this link.

http://www.voicesonic.com/panasonic/manuals/KX-TDE_100_200_600/Panasonic-KX-TDE100-200-600-Programming-Virtual-SIP-Card.pdf

There are a few PANASONIC manuaals here - look down to 600 range

http://www.voicesonic.com/customer/Panasonic-115-KX-TDE100-200-Manuals.html

I reckon the Panasonic KX-TDE 600 will be set up as a SIP trunk as if it is connecting to a Service provider and the CME will emulate the SP.

Regards

Alex

Regards, Alex. Please rate useful posts.

View solution in original post

paolo bevilacqua · ‎07-10-2012

Cisco CME is sending the requested authentication:

Proxy-Authorization: Digest username="22210960@bbtb.cyta.com.cy",realm="Registered Users",uri="sip:805@192.168.28.8:5060",response="887ec911f9a1c48a968c6ef7df7

After, in one case the PBX sends 'not found', in another, no repsose.

So, the problem is not caused by Cisco CME.

View solution in original post

Dmitri Popkov · ‎08-14-2012

Yea, sure.

On Panasonic I've phone nubmerx 2XX, on CCME - 5XX

Panasonic site (Configuration through PBX UMC)

1) Insert a V-IPGW16 card in your virtual unit (Item - 1.1)

2) Then go to properties of the system unit (righ click on V-IPGW16)

3) Click on gateway settings

4) Fill all fields in first line (specially - IP address in my config it's - 172.16.0.1 (and remember the number of gateway)). According to my example my gate number - "1"

5) Then click on D2IP and write in first column first nubmer of CISCO CME phone numbers, in my config it's - 5, second column next possible amount of numbers befor "5", my config - "2". And the last column - gate number - according to item 4) - gate number - "1"

7)Then item 2.6.1 on PBX UMC, choose third tab (numbers of local PBX). And for example in first line, write the first nubmer of phones, that belong to CME, according to my example, it's - 5.

That's all on Pana site

CCME 2901 (A bit easier than Pana)

---

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

----

voice class codec 1

codec preference 1 g711alaw

codec preference 2 g711ulaw

codec preference 3 g729br8

---

dial-peer voice 1000 voip

destination-pattern 2..

session target ipv4:172.16.0.250

dtmf-relay h245-alphanumeric

---

That's all h.323 - established successfully. Hope it will help you, If you have some difficulties, i'll help;)

Thanks

View solution in original post

acampbell · ‎02-10-2012

Hi,

Have a look at this link.

http://www.voicesonic.com/panasonic/manuals/KX-TDE_100_200_600/Panasonic-KX-TDE100-200-600-Programming-Virtual-SIP-Card.pdf

There are a few PANASONIC manuaals here - look down to 600 range

http://www.voicesonic.com/customer/Panasonic-115-KX-TDE100-200-Manuals.html

I reckon the Panasonic KX-TDE 600 will be set up as a SIP trunk as if it is connecting to a Service provider and the CME will emulate the SP.

Regards

Alex

Regards, Alex. Please rate useful posts.

chrysostomos1980 · ‎02-10-2012

Hi Alex again

we have a tunnel between two sites

from CME site is any special config that we have to do except dial peer?

Regards

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""

acampbell · ‎02-10-2012

Hi,

No as long as the IP routing from the CME to the PAN box is allowed I cant see a tunnel being an issue.

You will need to allow SIP to SIP & H323 internetworking if you have voice mail etc.

!

voice service voip

allow-connections sip to sip

allow-connections sip to h323

allow-connections h323 to sip

allow-connections h323 to h323

!

And may be SIP authentication along the lines of something like this

!
sip-ua
authentication username xyz password xyz realm cisco.com
!

HTH

Alex

Please rate useful posts

Regards, Alex. Please rate useful posts.

santiago · ‎07-10-2012

Hi Alex,

I'm doing the same case.

The issue is like this:

The Panasonic gets a call from the CME and asks for authentication, the CME doesn't sends and and the call is disconnected.

The only option to solve this is to register the Panasonic at the CME.

The CME will act as an SP ( will emulate SP ).

Can you explain me please how to do this?

Thanks a lot.

acampbell · ‎07-10-2012

Hi,

Here are a couple of links that may help you

http://www.cisco.com/en/US/docs/ios/12_3/sip/configuration/guide/trouble.html

http://www.cisco.com/en/US/products/sw/voicesw/ps4625/products_configuration_example09186a00808f9666.shtml#

Regards,
Alex.
Please rate useful posts.

Regards, Alex. Please rate useful posts.

Md. Saiful Islam Jehadi · ‎05-05-2017

It works for me. I added it in CUCM as h323 gateway and it works fine. bothway calls are working fine. I tested it by connecting through CUBE as well. that also works fine. Only the fact is i always considered it as H323 device.

paolo bevilacqua · ‎07-10-2012

There is no need to register CMe in order to support SIP digest authentication.

Just configure authentication under sip-ua.

santiago · ‎07-10-2012

Hi,

Thanks for replies :

Alex, i've looking for, but i've found that the CME has no possiblity to be SIP Proxy.

https://supportforums.cisco.com/thread/92668

Paolo,

See the attached wireshark: when the cisco sends a call to the Panasonic, the Panasonic ask for credentials ( That are configured at the panasonic ) and the cisco doesn't replies.

How i must set at the dial peer the credentials of this one?

http://www.2shared.com/file/HeKo3BK9/2-CYtoIL.html

Other fact : that same Panasonic is registered to an Asterisk with other Sip trunk and the calls there work bi-directional.

Thanks a lot,

paolo bevilacqua · ‎07-10-2012

post sip-ua configuration?

Also please do not post wireshark trace for these cases.

Post output of 'debug ccsip message' taken with 'term mon'.

santiago · ‎07-10-2012

Hi,

Sorry, have the ccsip trace:

As you will see, the Panasonic is asking for

SIP/2.0 407 Proxy Authentication Required

Where should i configure that for him?

Many Thanks again.

Sent:

INVITE sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECB191F

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

To: <805>

Date: Mon, 09 Jul 2012 10:36:10 GMT

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3131352065-3370652129-2590414196-2627930666

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1341830170

Contact: <0911>

Expires: 60

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 268

v=0

o=CiscoSystemsSIP-GW-UserAgent 2443 2038 IN IP4 192.168.5.5

s=SIP Call

c=IN IP4 192.168.5.5

t=0 0

m=audio 17260 RTP/AVP 8 101 19

c=IN IP4 192.168.5.5

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

001957: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECB191F

To: sip:805@192.168.28.8

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

CSeq: 101 INVITE

Timestamp: 1341830170

Content-Length: 0

001958: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECB191F

To: sip:805@192.168.28.8;tag=20079

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

CSeq: 101 INVITE

Allow: INVITE,ACK,CANCEL,BYE,REGISTER

Proxy-Authenticate: Digest realm="Registered Users",nonce="5ab56bd6ac58b061c3870e1c3871e3c6",algorithm=MD5

Content-Length: 0

001959: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ

001960: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:8A6B2308

001961: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECB191F

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

To: sip:805@192.168.28.8;tag=20079

Date: Mon, 09 Jul 2012 10:36:10 GMT

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

001962: Jul 9 10:36:10.830: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECC2404

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

To: sip:805@192.168.28.8

Date: Mon, 09 Jul 2012 10:36:10 GMT

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3131352065-3370652129-2590414196-2627930666

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 102 INVITE

Max-Forwards: 70

Timestamp: 1341830170

Contact: <0911>

Expires: 60

Allow-Events: telephone-event

Proxy-Authorization: Digest username="22210960@bbtb.cyta.com.cy",realm="Registered Users",uri="sip:805@192.168.28.8:5060",response="887ec911f9a1c48a968c6ef7df702bb8",nonce="5ab56bd6ac58b061c3870e1c3871e3c6",algorithm=MD5

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 268

v=0

o=CiscoSystemsSIP-GW-UserAgent 2443 2038 IN IP4 192.168.5.5

s=SIP Call

c=IN IP4 192.168.5.5

t=0 0

m=audio 17260 RTP/AVP 8 101 19

c=IN IP4 192.168.5.5

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

a=ptime:20

001963: Jul 9 10:36:11.022: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECC2404

To: sip:805@192.168.28.8

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

CSeq: 102 INVITE

Timestamp: 1341830170

Content-Length: 0

001964: Jul 9 10:36:11.022: //50729/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECC2404

To: sip:805@192.168.28.8;tag=27181

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

CSeq: 102 INVITE

Allow: INVITE,ACK,CANCEL,BYE,REGISTER

Content-Length: 0

001965: Jul 9 10:36:11.026: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

001966: Jul 9 10:36:11.026: //50729/BAA4A4019A66/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SPI_EVENT

001967: Jul 9 10:36:11.026: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP

001968: Jul 9 10:36:11.030: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_DNS_RESOLVE

001969: Jul 9 10:36:11.030: //50730/BAA4A4019A66/SIP/Event/Session-Timer/sipSTSLMain: Event: E_STSL_SESSION_REFRESH_REQ

001970: Jul 9 10:36:11.030: //50730/BAA4A4019A66/SIP/Event/Session-Timer/sipSTSLMain: dir:1, method:102, resp_code:0, container:8A6B36F8

001971: Jul 9 10:36:11.034: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.5.5:5060;branch=z9hG4bKBECC2404

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16A30-1EEB

To: sip:805@192.168.28.8;tag=27181

Date: Mon, 09 Jul 2012 10:36:10 GMT

Call-ID: BBCAD501-C8E811E1-9A6B9574-9CA30A2A@192.168.5.5

Max-Forwards: 70

CSeq: 102 ACK

Allow-Events: telephone-event

Content-Length: 0

001972: Jul 9 10:36:11.034: //50730/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.253:5060;branch=z9hG4bKBECD1B36

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

To: <805>

Date: Mon, 09 Jul 2012 10:36:11 GMT

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 3131352065-3370652129-2590414196-2627930666

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Max-Forwards: 70

Timestamp: 1341830171

Contact: <0911>

Expires: 60

Allow-Events: telephone-event

Content-Type: application/sdp

Content-Disposition: session;handling=required

Content-Length: 337

v=0

o=CiscoSystemsSIP-GW-UserAgent 8935 6474 IN IP4 192.168.10.253

s=SIP Call

c=IN IP4 192.168.10.253

t=0 0

m=audio 18790 RTP/AVP 8 0 18 101 19

c=IN IP4 192.168.10.253

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=yes

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=rtpmap:19 CN/8000

001973: Jul 9 10:36:11.134: //50730/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.10.253:5060;received=10.226.95.253;branch=z9hG4bKBECD1B36

To: <805>;tag=h7g4Esbg_775112800-1341830299740

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

CSeq: 101 INVITE

Contact:

Record-Route: <10.224.42.132>

Content-Type: application/sdp

Content-Length: 222

Session: Media

P-Early-Media: sendonly

Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO

v=0

o=BroadWorks 19134227 1 IN IP4 10.224.42.132

s=-

c=IN IP4 10.224.42.7

t=0 0

m=audio 14492 RTP/AVP 8 101

c=IN IP4 10.224.42.7

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

001974: Jul 9 10:36:15.910: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT

001975: Jul 9 10:36:15.914: //50730/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Sent:

CANCEL sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.253:5060;branch=z9hG4bKBECD1B36

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

To: <805>

Date: Mon, 09 Jul 2012 10:36:11 GMT

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

CSeq: 101 CANCEL

Max-Forwards: 70

Timestamp: 1341830175

Reason: Q.850;cause=16

Content-Length: 0

001976: Jul 9 10:36:15.942: //50730/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.10.253:5060;received=10.226.95.253;branch=z9hG4bKBECD1B36

To: <805>;tag=h7g4Esbg_775112800-1341830299740

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

CSeq: 101 CANCEL

Content-Length: 0

001977: Jul 9 10:36:15.962: //50730/BAA4A4019A66/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 487 Request terminated

Via: SIP/2.0/UDP 192.168.10.253:5060;received=10.226.95.253;branch=z9hG4bKBECD1B36

To: <805>;tag=h7g4Esbg_775112800-1341830299740

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

CSeq: 101 INVITE

Content-Length: 0

001978: Jul 9 10:36:15.962: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:805@192.168.28.8:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.10.253:5060;branch=z9hG4bKBECD1B36

From: "PavlosG" <>0911@bbtb.cyta.com.cy>;tag=C2F16BB4-1564

To: <805>;tag=h7g4Esbg_775112800-1341830299740

Date: Mon, 09 Jul 2012 10:36:11 GMT

Call-ID: BC060A82-C8E811E1-9A6D9574-9CA30A2A@192.168.5.5

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

paolo bevilacqua · ‎07-10-2012

Cisco CME is sending the requested authentication:

Proxy-Authorization: Digest username="22210960@bbtb.cyta.com.cy",realm="Registered Users",uri="sip:805@192.168.28.8:5060",response="887ec911f9a1c48a968c6ef7df7

After, in one case the PBX sends 'not found', in another, no repsose.

So, the problem is not caused by Cisco CME.

santiago · ‎07-10-2012

Mmm, now it suprise me.

Where is set :

Proxy-Authorization: Digest username="

22210960@bbtb.cyta.com.cy

",realm="Registered Users",

I guess here:

no dial-peer outbound status-check pots

sip-ua

credentials username 22210960 password 7 XXXXX realm bxx.cyta.com.cy

authentication username 22210960@bbtb.cyta.com.cy password 7 XXXXX

no remote-party-id

retry invite 2

retry register 3

retry options 2

timers expires 60000

timers connect 100

timers register 1000

registrar dns:bbtb.cyta.com.cy expires 110

sip-server dns:bbtb.cyta.com.cy

host-registrar

!

But where the realm "Registered Users" comes from ? I didn't found it.

Can i set this credentials for each specific dial-peer?

Your help is most valuated for me.

paolo bevilacqua · ‎07-10-2012

Cisco CME doesn't really give much importance to realm. it will respond anyway with the authentication configured under sip-ua, maybe try configuring the realm anyway.

Unfortunatly is not possible to configure multiple authentication credentials for different DPs. Only a single one under sip-ua.

chrysostomos1980 · ‎07-10-2012

Hi to all

This tread opened by me

Unfortunately the news are not good

After one month troubleshooting with TAC we didn't manage to solve the problem because the problem with the authentication coming from panasonic

So we have established only incoming calls but not outgoing to panasonic

Sorry guys for the bad news

Please rate all useful posts Regards Chrysostomos ""The Most Successful People Are Those Who Are Good At Plan B""