Cisco CP-8831 phone and Asterisk

Asterisk Asterisk Support
1

Dear all,

I need to register the Cisco Unified IP Conference Phone 8831 phone (cisco.com/c/en/us/products/c … 26887.html) with our Asterisk server (version 1.8.20.0).

The preloaded firmware in the said phone is SIP enabled and I managed to get the CNF.XML for this specific phone model from a different Cisco Unified CM server. Since I can’t attach a file to this post I will paste the contents into the second post - can someone point me which options I need to set for the phone to register at our Asterisk installation?

Basically I need to set the following:

  1. IP of the Asterisk sever (sipIpAddr1 in the XML?)
  2. Login (userId in the XML?)
  3. Password-secret (can’t find this in the XML - maybe PhonePassword?)

Ports are SIP default.

Thanks in advance to all for your help, I spent the last week trying to figure this out but nothing :frowning:

Jan

PS: The phone itself doesen’t have a WebUI where you can set anything - only info about current status. Thus, the only way to get the XML config is from an existing CUCM server installation - terrible solution by Cisco…!

2

Contents of the XML configuration file for the CP-8831:

<?xml version="1.0" encoding="UTF-8"?> true *** SIP admin admin 0 true true false false 0 0 2013i tzupdater.jar 000000 Off Disabled false 0 Default CMLocal M/D/Y Greenwich Standard Time Etc/GMT Default true *** *** 2000 5060 5061 2427 2428 *** Disable Disable false 2000 2000 2000 *** 5060 5060 5060 false 120 2445 *** USECALLMANAGER 5060 USECALLMANAGER 5060 USECALLMANAGER 5060 true true x-cisco-serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 1 true true 2 2 0 true false 6 10 180 3600 5 120 120 5 500 4000 70 true None 1 false true false false none 101 3 avt 3 2 true 15000 15000 10 true false 16384 32766 5060 184 180 176 164 168 168 156 148 140 132 132 136 136 128 128 0 USb0ec918f-b9ee-994b-57ae-345883c1fde8.xml false false 0 true 2 sip8831.9-3-3-5 0105000800110 1399489720-6218796c-39dd-41a1-9115-4c816a263ebf English_United_States 1 en_US iso-8859-1 United_States United_States 64 1 0 *** 96 0 96 4 5 1 0 0 0 false 0 0 0 3804 *** false 1 *81 *82 *83 *84 *85 3600 *** *** 2 Missed Calls Application:Cisco/MissedCalls Voicemail Application:Cisco/Voicemail Received Calls Application:Cisco/ReceivedCalls Placed Calls Application:Cisco/PlacedCalls Personal Directory Application:Cisco/PersonalDirectory Corporate Directory Application:Cisco/CorporateDirectory
3

UPDATE - Still no go… tried messing around with the settings - after maybe 50 TFTP uploads with different settings… still no connection - the phone doesen’t even try to connect to Asterisk (nothing in the Asterisk CLI log)

HEEEEEELP PLEASE

4

Did you change “nat”=“No” in the extension setting?

[code]<?xml version="1.0" ?>

SIP
root
cisco


M/D/YA
Saudi Arabia Standard Time



Unicast







192.168.1.1 <?// change it to asterisk ip //?>

5060










true


192.168.1.10 <?// change it to asterisk ip //?>
5060

none
General <?// Display name //?>

		<line button="1">
			<featureID>9</featureID>
			<featureLabel>8001</featureLabel>
			<proxy>USECALLMANAGER</proxy>	<?// must be USECALLMANAGER //?>
			<port>5060</port>
			<name>8001</name>
			<authName>8001</authName> <?// extension username //?>
			<authPassword>###</authPassword> <?// extension password  //?>
			<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
			<messagesNumber>*97</messagesNumber>
		</line>

	</sipLines>
	<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>

<networkLocale>United_States</networkLocale><networkLocaleInfo><name>United_States</name><uid>64</uid><version>1.0.0.0-1</version></networkLocaleInfo>
[/code]