SIP phone with CCME

Answered Question
May 30th, 2007

Hi,

I'm using CCME 4.1 and now I had purchased a linksys SIP IP Phone (model SPA921) and want to add this to my CCME. So what configuration should I have to do?

anyone has configured this and using it?

Regards

I have this problem too.
0 votes
Correct Answer by derek.small about 9 years 5 months ago

Here is the parts that you need to get the phone to register.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

h323

call start slow

modem passthrough nse codec g711ulaw

sip

registrar server expires max 240 min 60

!

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g729br8

!

voice class codec 2

codec preference 2 g729r8

codec preference 3 g729br8

!

voice class codec 3

codec preference 1 g711ulaw

!

!

!

voice class h323 1

h225 timeout tcp establish 3

!

!

voice register global

mode cme

source-address 10.72.13.19 port 5060

max-dn 200

max-pool 20

timezone 13

tftp-path slot0:

create profile sync 0390651099874129

ntp-server 10.71.13.19 mode directedbroadcast

!

voice register dn 1

number 4020

allow watch

name User1

label User1

!

voice register dn 2

number 4021

allow watch

name User2

label User2

!

voice register dn 3

number 4022

allow watch

name LinkSys

label LinkSys

!

voice register dn 4

number 4050

allow watch

name CIN-Tandberg

label CIN-Tandberg

!

voice register template 1

session-transport udp

softkeys hold Resume Newcall

softkeys idle Newcall Redial Cfwdall

softkeys connected Endcall Trnsfer Confrn Hold

voicemail 4200 timeout 30

!

voice register dialplan 1

type 7940-7960-others

pattern 1 4...

pattern 2 ....

pattern 3 .

!

voice register pool 1

id mac 0004.F213.2465

type P600

number 1 dn 2

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description SoundPoint501

!

voice register pool 2

id mac 0004.F210.C66B

type P600

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description SoundPoint301

!

voice register pool 3

id mac 000E.08DE.1C9B

number 1 dn 3

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description LinkSys-SPA942

!

voice register pool 4

id mac 0050.6001.1419

number 1 dn 4

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description CIN-Tandberg

!

!

!

!sip-ua

retry options 0

mwi-server ipv4:10.72.13.20 expires 3600 port 5060 transport udp

registrar ipv4:10.72.13.20 expires 3600 secondary

sip-server ipv4:10.72.13.20

presence enable

refer-ood enable 50

Correct Answer by Paolo Bevilacqua about 9 years 5 months ago

Please configure

voice register pool 10

number 1 dn 1

no application sip.app

Also you may need to make the username same as then number you want to register.

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Overall Rating: 5 (6 ratings)
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Paolo Bevilacqua Thu, 05/31/2007 - 17:23

voice register pool xx

mac 0000.0000.0000

username 111 password 1234

In the spa921 configure the same username and password. For simplicity, use extension number as username.

Hope this helps, please rate post if it does!

rv_viji Fri, 06/01/2007 - 21:39

Hi,

Where should I configure the username in the SPA921, I do not find any option on any of the menus in the phone...

Regards

Paolo Bevilacqua Sat, 06/02/2007 - 08:03

Hello,

I could not locate the administrator guide for the spa921, but I see in the data sheet that digest authentication is supported, plus we know that other users have been successful in doing this, so you should be able to do that as well!

rv_viji Tue, 06/26/2007 - 21:30

Hi,

I have done the settings on the phone as per the instructions in the guide, but still the phone is not getting registered with my CCME.

I think I'm going wrong somewhere in the CCME configuration...

Anybody has any sample configuration of CCME for SIP phones....

It will be great help if anybody can share the same...

Regards

Paolo Bevilacqua Wed, 06/27/2007 - 01:51

1. voice register dn number same as number on the phone

2. voice register pool username and password to match what is configured in the phone, suggest it is the number itself.

3. realm none, or same on both voice register global and phone.

All failing, includ here configuration and output of "term mon" and "debug ccsip message".

rv_viji Wed, 06/27/2007 - 03:21

Configuration:

voice service voip

allow-connections sip to sip

sip

registrar server expires max 1200 min 300

!

voice register global

mode cme

max-dn 288

max-pool 96

!

voice register dn 1

number 500

!

voice register pool 10

id mac 000E.08DD.8B04

application sip.app

username bitlink password bitlink

!

Debug output:

Jun 27 11:38:53.555: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

REGISTER sip:192.168.2.11 SIP/2.0

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-f5e2cb53

From: ;tag=28f185c25429a4eo0

To:

Call-ID: [email protected]

CSeq: 65293 REGISTER

Max-Forwards: 70

Contact: ;expires=3600

User-Agent: Sipura/SPA921-4.1.10(b)

Content-Length: 0

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

Jun 27 11:38:53.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-f5e2cb53

From: ;tag=28f185c25429a4eo0

To:

Date: Wed, 27 Jun 2007 11:38:53 GMT

Call-ID: [email protected]

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 65293 REGISTER

Content-Length: 0

Jun 27 11:38:53.559: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-f5e2cb53

From: ;tag=28f185c25429a4eo0

To: ;tag=56904-C4A

Date: Wed, 27 Jun 2007 11:38:53 GMT

Call-ID: [email protected]

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 65293 REGISTER

Content-Length: 0

Correct Answer
Paolo Bevilacqua Wed, 06/27/2007 - 03:30

Please configure

voice register pool 10

number 1 dn 1

no application sip.app

Also you may need to make the username same as then number you want to register.

rv_viji Wed, 06/27/2007 - 03:50

Hey,

Many thanks now its registered with my CCME..

I changed the username to the phone number then it got registered...

But now I'm able to place a call from my Cisco IP Phone to this SIP IP Phone, however the vice verse is not happening....

What could be the problem...

Attached the debug ccsip output taken during a call being placed from SIP IP Phone to Cisco IP Phone..

Jun 27 12:11:07.627: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-4438c165

From: ;tag=e3176a26cbbe8975o0

To:

Call-ID: [email protected]

CSeq: 101 INVITE

Max-Forwards: 70

Contact:

Expires: 240

User-Agent: Sipura/SPA921-4.1.10(b)

Content-Length: 393

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

Content-Type: application/sdp

v=0

o=- 76516 76516 IN IP4 192.168.2.34

s=-

c=IN IP4 192.168.2.34

t=0 0

m=audio 16456 RTP/AVP 0 2 4 8 18 96 97 98 101

a=rtpmap:0 PCMU/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729a/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:97 G726-24/8000

a=rtpmap:98 G726-16/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:30

a=sendrecv

Jun 27 12:11:07.631: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

SIP/2.0 488 Not Acceptable Media

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-4438c165

From: ;tag=e3176a26cbbe8975o0

To: ;tag=22EC00-9DE

Date: Wed, 27 Jun 2007 12:11:07 GMT

Call-ID: [email protected]

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 101 INVITE

Allow-Events: telephone-event

Reason: Q.850;cause=65

Content-Length: 0

Jun 27 12:11:07.643: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK-4438c165

From: ;tag=e3176a26cbbe8975o0

To: ;tag=22EC00-9DE

Call-ID: [email protected]

CSeq: 101 ACK

Max-Forwards: 70

Contact:

User-Agent: Sipura/SPA921-4.1.10(b)

Content-Length: 0

Paolo Bevilacqua Wed, 06/27/2007 - 05:45

Hi,

which IP Phone and firmware are you using?

It could be the IP phone that is refusing the call for some related to the codec.

derek.small Wed, 06/27/2007 - 08:29

Looks like you have a codec mis-match. Try a voice-class that includes at least G711, G729r8, and G729br8. You should also check your "voice register dialplan"

rsefer Tue, 07/17/2007 - 14:52

Hello,

I have do the same config

In my case

CCME and Linksys are not in the same LAN

And IP adreses are not real (NAT)

Following debug messages

Sent:

SIP/2.0 400 Bad Request - 'Invalid IP Address'

Via: SIP/2.0/UDP 88.230.21.105:5060;branch=z9hG4bK-adf53327

From: 712 ;tag=86ce9cf293dced4ao0

To: 712 ;tag=234B8A4-FAF

Date: Wed, 18 Jul 2007 01:46:40 GMT

Call-ID: [email protected]

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 21737 REGISTER

Content-Length: 0

THANKS

Paolo Bevilacqua Tue, 07/17/2007 - 15:04

Hi,

the error is because source and destination of the request are the same. Please include a diagram of all devices involved, how they are connected, addresses and configuration.

derek.small Wed, 06/27/2007 - 08:25

Just posted my config to another thread, but it works, however still haven't gotten MWI to work, and when I hit the messages button, it's treating the call like CFNA/CFB to CUE. If I dial my voice mail extension directly it works fine.

Also I'm running 2.0.3 on my SP501 phone, and CME 4.1.0.1 (IOS 12.4(11)XJ3 on a 3725).

rv_viji Thu, 06/28/2007 - 08:29

Hi,

Can you post your config to this thread..

Regards

Paolo Bevilacqua Thu, 06/28/2007 - 08:47

Do you have 12.4(11)XJ3 and if not, can you upgrade to that ?

I see no reason for the "media not acceptable" message as the codecs specified are right. Perhaps, in the linksys config, try to exclude all the codecs but g.711u.

Correct Answer
derek.small Thu, 06/28/2007 - 10:43

Here is the parts that you need to get the phone to register.

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

redirect ip2ip

h323

call start slow

modem passthrough nse codec g711ulaw

sip

registrar server expires max 240 min 60

!

!

voice class codec 1

codec preference 1 g711ulaw

codec preference 2 g729r8

codec preference 3 g729br8

!

voice class codec 2

codec preference 2 g729r8

codec preference 3 g729br8

!

voice class codec 3

codec preference 1 g711ulaw

!

!

!

voice class h323 1

h225 timeout tcp establish 3

!

!

voice register global

mode cme

source-address 10.72.13.19 port 5060

max-dn 200

max-pool 20

timezone 13

tftp-path slot0:

create profile sync 0390651099874129

ntp-server 10.71.13.19 mode directedbroadcast

!

voice register dn 1

number 4020

allow watch

name User1

label User1

!

voice register dn 2

number 4021

allow watch

name User2

label User2

!

voice register dn 3

number 4022

allow watch

name LinkSys

label LinkSys

!

voice register dn 4

number 4050

allow watch

name CIN-Tandberg

label CIN-Tandberg

!

voice register template 1

session-transport udp

softkeys hold Resume Newcall

softkeys idle Newcall Redial Cfwdall

softkeys connected Endcall Trnsfer Confrn Hold

voicemail 4200 timeout 30

!

voice register dialplan 1

type 7940-7960-others

pattern 1 4...

pattern 2 ....

pattern 3 .

!

voice register pool 1

id mac 0004.F213.2465

type P600

number 1 dn 2

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description SoundPoint501

!

voice register pool 2

id mac 0004.F210.C66B

type P600

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description SoundPoint301

!

voice register pool 3

id mac 000E.08DE.1C9B

number 1 dn 3

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description LinkSys-SPA942

!

voice register pool 4

id mac 0050.6001.1419

number 1 dn 4

template 1

dialplan 1

dtmf-relay rtp-nte

voice-class codec 1

description CIN-Tandberg

!

!

!

!sip-ua

retry options 0

mwi-server ipv4:10.72.13.20 expires 3600 port 5060 transport udp

registrar ipv4:10.72.13.20 expires 3600 secondary

sip-server ipv4:10.72.13.20

presence enable

refer-ood enable 50

rv_viji Fri, 06/29/2007 - 22:33

Hi,

Thanks, it worked after I configured the voice class codec...

Thanks to all for your support.

Have a great day..

Regards

Paolo Bevilacqua Sat, 06/30/2007 - 03:27

Good catch by derek, I've rated his post!

I was comparing the trace with another SIP phone that works ok w/o the need of configuring voice class, and could not see an obvious difference.

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