Hi again,

 

I actually reloaded my router with clean configuration and then
re-configured CME, however I am still seeing the same problem.  I erased the
configurations on the phones before this all happened, so I assume this is
maybe part of the problem.  I don’t know why it would be though, as the
phones are getting IP addresses from DHCP and communicating with CME.

 

This is my relevant configs:

 

 

< ----- DHCP FOR PHONES ----- >

 

 

ip dhcp excluded-address 10.5.202.1

ip dhcp pool SC_PHONES

   network 10.5.202.0 255.255.255.0

   option 150 ip 10.5.202.1 

   default-router 10.5.202.1

 

 

< -----  VOICE SERVICE ----- >

 

voice service voip 

 allow-connections sip to sip

 fax protocol cisco 

 sip

  bind control source-interface Vlan250

  bind media source-interface Vlan250

  registrar server expires max 1200 min 500

 

< ----- VOICE CODEC ----- >

 

 

voice class codec 1

 codec preference 1 g711alaw

 codec preference 2 g711ulaw

 codec preference 3 g729r8

 

 

< ----- SIP CME CONFIG ----- >

 

voice register global

 mode cme

 source-address 10.5.202.1 port 5060

 max-dn 20

 max-pool 2

 load 7945 SIP45.9-0-3S

 load 7942 SIP42.9-0-3S

 authenticate register

 date-format Y/M/D

 voicemail 4500

 url directory http://10.5.202.1/localdirectory

 create profile sync 0001302544054013

 ntp-server 10.5.200.1 mode directedbroadcast

 

< ----- PHONE 1 LINE 1 ----- >

 

voice register dn  1

 number 4001

 call-forward b2bua busy 4500  

 call-forward b2bua noan 4500 timeout 10

 allow watch

 name Site C Phone 1

 label 4001

 

< ----- PHONE 2 LINE 1 ----- >

 

voice register dn  2

 number 4002

 call-forward b2bua busy 4500  

 call-forward b2bua noan 4500 timeout 10

 allow watch

 name Site C Phone 2

 label 4002

 

 

< ----- PHONE 1 (7942) ----- >

 

voice register pool  1

 id mac 0024.9733.6C28

 type 7942

 number 1 dn 1

 presence call-list

 dtmf-relay rtp-nte sip-notify

 voice-class codec 1

 username scuser1 password cisco

 description +442321314001

 blf-speed-dial 1 4002 label "SCPH2 4002" device

 privacy off

 

 

< ----- PHONE 2 (7945) ----- >

 

voice register pool  2

 id mac 0024.14B2.F542

 type 7945

 number 1 dn 2

 presence call-list

 dtmf-relay rtp-nte sip-notify

 voice-class codec 1

 username scuser2 password cisco

 description +442321314002

 blf-speed-dial 1 4001 label "SCPH1 4001" device

 privacy off

 

< ----- TFTP FILES ----- > 

 

tftp-server flash:SIP/apps42.9-0-3TH1-22.sbn alias apps42.9-0-3TH1-22.sbn

tftp-server flash:apps42.9-0-3TH1-22.sbn alias cnu42.9-0-3TH1-22.sbn

tftp-server flash:SIP/cvm42sip.9-0-3TH1-22.sbn alias
cvm42sip.9-0-3TH1-22.sbn

tftp-server flash:SIP/dsp42.9-0-3TH1-22.sbn alias dsp42.9-0-3TH1-22.sbn

tftp-server flash:SIP/jar42sip.9-0-3TH1-22.sbn alias
jar42sip.9-0-3TH1-22.sbn

tftp-server flash:SIP/SIP42.9-0-3S.loads alias SIP42.9-0-3S.loads

tftp-server flash:SIP/term42.default.loads alias term42.default.loads

tftp-server flash:SIP/apps45.9-0-3TH1-22.sbn alias apps45.9-0-3TH1-22.sbn

tftp-server flash:SIP/cnu45.9-0-3TH1-22.sbn alias cnu45.9-0-3TH1-22.sbn

tftp-server flash:SIP/cvm45sip.9-0-3TH1-22.sbn alias
cvm45sip.9-0-3TH1-22.sbn

tftp-server flash:SIP/dsp45.9-0-3TH1-22.sbn alias dsp45.9-0-3TH1-22.sbn

tftp-server flash:SIP/jar45sip.9-0-3TH1-22.sbn alias
jar45sip.9-0-3TH1-22.sbn

tftp-server flash:SIP/SIP45.9-0-3S.loads alias SIP45.9-0-3S.loads

tftp-server flash:SIP/term45.default.loads alias term45.default.loads

 

< ----- PHONE PORTS ---à

 

interface FastEthernet0/2/2

 switchport access vlan 150

 switchport voice vlan 250

 spanning-tree portfast

!

interface FastEthernet0/2/3

 switchport access vlan 150

 switchport voice vlan 250

 spanning-tree portfast

 

I am still seeing the phone say unprovisioned.  As you suggested I looked at
the show voice register tftp command and I can see the SEP<mac>.cnf.xml
statements for the phones.

 

< -----  show voice register tftp ----- >

 

R3#show voice register tftp

tftp-server syncinfo.xml url system:/cme/sipphone/syncinfo.xml

tftp-server SIPDefault.cnf url system:/cme/sipphone/SIPDefault.cnf

tftp-server softkeyDefault_kpml.xml url
system:/cme/sipphone/softkeyDefault_kpml.xml

tftp-server softkeyDefault.xml url system:/cme/sipphone/softkeyDefault.xml

tftp-server SEP002497336C28.cnf.xml url
system:/cme/sipphone/SEP002497336C28.cnf.xml

tftp-server SEP002414B2F542.cnf.xml url
system:/cme/sipphone/SEP002414B2F542.cnf.xml

 

Also, when I look at the status messages on the phone, I see a “Error
Verifying Config Info” message.

 

Any help is appreciated,

 

Jeff

 

 

From: Daniel Berlinski [mailto:dberlin...@gmail.com] 
Sent: Friday, October 01, 2010 4:20 PM
To: CCIE Voice GMAIL
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME

 


Make sure after each change you commit to your SIP phones configuration
files you do a create profile under voice reg global.  Verify you have your
config file ready for download by issueing show voice register tftp and
check the mac address of your SIP phone is in the list.

A couple of other things to note:
You need IP connectivity to your CME router.
If your phones are remote to you you need to bind the SIP interface you are
sourcing your SIP packets and use that IP address as your source address
under voice register global.
Authenticate register is a must also if your phones are remote to your SIP
CME router.
Lastly ensure your voice reg pools have username and password and that you
have a number assigned to your phones as the primary number.

After all these create profile and if still with troubles please post your
configs.



Cheers

On Sat, Oct 2, 2010 at 12:08 PM, CCIE Voice GMAIL
<givemeccievoice2...@gmail.com> wrote:

Hi Dan,

 

Thanks for the response.

 

Before I could test what you had told me, I must have screwed something up.
I keep getting a message on both of my CME phones saying “Unprovisioned”.  I
have reloaded my router and re-configured everything again but I am still
getting that message.  

 

Has anyone seen this before?

 

Jeff

 

From: Daniel Berlinski [mailto:dberlin...@gmail.com] 
Sent: Friday, October 01, 2010 2:55 PM
To: CCIE Voice GMAIL
Cc: osl osl
Subject: Re: [OSL | CCIE_Voice] SIP Phones in CME

 

description in voice register pool config mode.

On Sat, Oct 2, 2010 at 10:51 AM, CCIE Voice GMAIL
<givemeccievoice2...@gmail.com> wrote:

Hi everyone,

 

I am having a hard time remembering what command will affect the number
displayed in the upper-right of the phones for CME.  

 

With SCCP, I know the description command will effect that number.  How do
you change this value for SIP phones registered to CME?

 

Thanks for the help,

 

Jeff

 


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_______________________________________________
For more information regarding industry leading CCIE Lab training, please
visit www.ipexpert.com

 

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