Please provide assistance registering a Cisco 8961

Hi, I have a cisco 8961 knocking around with the SIP firmware already added. I have a cisco POE switch and a PC running FreePBX. I’ve tried to put the XML file in TFTP, but the TFTP isn’t showing the phone trying to pull it. The extenstion I have for it is 250 and the secret is d7c15c4b34df3d9fc2dd41384951f74d. (and yes I’m going to change it after this). After all of this, the phone sit while displaying “Phone Not Registered” Can someone please help me with this issue, I’ve always wanted one of these systems after we learned about VOIP in Cisco academy. Below is my XML file i’m trying to upload to the 8961.

<?xml version="1.0"?>

-

SIP

admin

password

-

-

M/D/YA

Pacific Standard/Daylight Time

-

-

pool.ntp.org

Unicast

-

-

-

-

-

2000

5060

5061

192.168.0.16

-

-

5060

true

-

true

x-serviceuri-cfwdall

x-cisco-serviceuri-pickup

x-cisco-serviceuri-opickup

x-cisco-serviceuri-gpickup

x-cisco-serviceuri-meetme

x-cisco-serviceuri-abbrdial

false

2

true

true

2

2

0

true

-

6

10

180

3600

5

120

120

5

500

4000

70

false

None

1

false

true

false

false

101

3

avt

false

false

3

Datamaster

1

false

10

false

-

-

9

8961

USECALLMANAGER

5060

250

250

-

2

3

250

e13cd3c19f5b7f89688c04bd7b40bfae

false

1

*97

4

5

250

-

true

false

false

true

5060

16348

20134

184

0

dialplan.xml

-

true

2

SIP45.8-4-2S

-

false

false

0

1

0

0

0

0

1

1,2,3,4,5,6,7

00:00

00:00

00:00

1

1

-

en_US

1.0.0.0-1

iso-8859-1

-

1.0.0.0-1

1

96

0

96

4

0

-

-

3804

false

If the phone isn’t pulling your config file from the TFTP directory, your Option 66 in the DHCP server is probably not set correctly.

Be warned that this phone will not work the way you learned in Cisco Academy when installed as a SIP phone in Asterisk without patches to Asterisk. In general, Cisco phones in SIP mode are among the hardest phones in the Internet to get working.

How would I change option number 66? This won’t be used as an academy project, just a proof of concept and a hobby. I already have some other SIP phones hooked up and wanted a challenge.

You need to modify the DHCP config file for your specific DHCP server. That’s going to vary with the DHCP server you are using.

In general, the steps are:

  1. Define a new Option 66 with the correct payload type (typ: “ip-address”) and call it “Cisco-tftp-server” or similar.
  2. In the group/phone definition, set the Named Option 66 to whatever you need to make the phone go to your VOIP Server (wherever you are keeping the TFTP files).

That didn’t work. phone still says not registered

Without logs, there’s no way of knowing what the problem is. Check the /var/log/asterisk/full log file and tell us what the system is saying about the registration.

There does not seem to be anything in the logs about the phone attempting to register

If the phone isn’t trying to register, that means that the address for the server in the config XML file isn’t right or that the phone is locked out by the firewall. Check with tcpdump and see if the phone is trying to register, but the integrated firewall has you locked out.

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