Asterisk with cisco 7941 not work

Hi guys,

I need some help, I’m not able to register my phone 7941G in asterisk version 1.8.13.1 , my SEPXXXX.cnf.xml file contains the following code :

<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/Y</dateTemplate>
<timeZone>Central European Time</timeZone>
<ntps>
<ntp>
<name>172.16.0.50</name>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
	<callManager>
	<ports>
		<ethernetPhonePort>2000</ethernetPhonePort>
		<sipPort>5060</sipPort>
		<securedSipPort>5061</securedSipPort>
	</ports>
<processNodeName>172.16.0.50</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<enableVad>false</enableVad>
<preferredCodec>g711ulaw</preferredCodec>
<natEnabled>false</natEnabled>
<phoneLabel>Cisco_Phone</phoneLabel>
	<sipLines>
		<line button="1">
		<featureID>9</featureID>
		<featureLabel>9002</featureLabel>
		<proxy>172.16.0.50</proxy>
		<name>9002</name>
		<displayName>9002</displayName>
		<authName>9002</authName>
		<authPassword>123mudar</authPassword>
		<messagesNumber>9002</messagesNumber>
</line>
		<line button="2">
		<featureID>21</featureID>
		<featureLabel>SpeedDial</featureLabel>
		<speedDialNumber>80808080</speedDialNumber>
		</line>
</sipLines>
<dialTemplate>dialplan.xml</dialTemplate>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
</commonProfile>
<loadInformation>SIP41.8-2-2SR1S</loadInformation>
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp>
<directoryURL></directoryURL>
<servicesURL></servicesURL>
</device> 

I boot the phone with the sequence 123456789 * 0 # and it loads all the files from the TFTP , then I access to the phone URL and he introduced the following error:

WRN 21:20:37.396743 JVM: Thread-3|cip.l10n.NetworkLocaleProperty:? - Unable to process LocaleProperty 'device.settings.config.localization.networklocale'
NOT 21:20:38.863523 JVM: Thread-3|cip.sipcc.c:  - initializeLinePlane(): Mgmt Interface is in Service now..
NOT 21:20:39.153626 tftpClient: tftp request rcv'd from /usr/tmp/tftp, srcFile = dialplan.xml, dstFile = /usr/cache/DP817620086 
NOT 21:20:39.156587 tftpClient: auth server - tftpList[0] = 172.16.0.11 
NOT 21:20:39.157382 tftpClient: look up server - 0 
WRN 21:20:39.160269 SECD: WARN:lookupCTL: ** no CTL, assume TFTP 172.16.0.11 NONSECURE
NOT 21:20:39.164441 tftpClient: secVal = 0xa 
NOT 21:20:39.165288 tftpClient: 172.16.0.11 is a NONsecure server 
NOT 21:20:39.166034 tftpClient: temp retval = SRVR_NONSECURE, keep looking 
NOT 21:20:39.166747 tftpClient: retval = 10 
NOT 21:20:39.167478 tftpClient: Secure file requested 
NOT 21:20:39.168184 tftpClient: Non secure file approved  -- dialplan.xml  
NOT 21:20:39.183457 TFTP: [17]:Requesting dialplan.xml from 172.16.0.11 
NOT 21:20:39.194703 TFTP: [17]:Finished --> rcvd 90 bytes 
ERR 21:21:11.100842 JVM: %REG auth failed: ack timer
ERR 21:21:43.125804 JVM: %REG auth failed: ack timer
ERR 21:22:15.135824 JVM: %REG auth failed: ack timer
ERR 21:22:47.150861 JVM: %REG auth failed: ack timer
ERR 21:23:19.160881 JVM: %REG auth failed: ack timer
ERR 21:23:51.165921 JVM: %REG auth failed: ack timer
ERR 21:24:23.170825 JVM: %REG auth failed: ack timer
ERR 21:24:55.170916 JVM: %REG auth failed: ack timer
ERR 21:25:27.181088 JVM: %REG auth failed: ack timer
ERR 21:25:59.190840 JVM: %REG auth failed: ack timer
ERR 21:26:31.200874 JVM: %REG auth failed: ack timer
ERR 21:27:03.210801 JVM: %REG auth failed: ack timer

The phones are only to be used internally with private IPs , I do not need NAT to make calls out of my internal network

I can register a zoiper phone and x-lite and talk in the asterisk, but this hardphone not, why is wrong?

Thanks in advance

Folks, some help where, please…

Anyone have cisco ip phone with asterisk???

Hell again,

Excuse the forum to be me to answer myself , but I’m hoping to get overcome my problem and someone has been kind enough to help me, so far already got the Cisco phone appears on the asterisk , but line 9002 is unreachable , and it does not work , although it can record , does anyone know why?

A capture with wireshark said that:

[quote]OPTIONS sip:9002@172.16.0.100:5060;transport=udp SIP/2.0•
Via: SIP/2.0/UDP 172.16.0.50:5060;branch=z9hG4bK0b53edb0;rport•
Max-Forwards: 70•
From: “asterisk” sip:asterisk@172.16.0.50;tag=as5dbbcede•
To: sip:9002@172.16.0.100:5060;transport=udp
Contact: sip:asterisk@172.16.0.50:5060
Call-ID: 7d75f1fd39f2e8815a2772ef1219aa04@172.16.0.50:5060•
CSeq: 102 OPTIONS•
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3•
Date: Wed, 17 Jun 2015 16:02:52 GMT•
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH•
Supported: replaces, timer•
Content-Length: 0•

OPTIONS sip:9002@172.16.0.100:5060;transport=udp SIP/2.0•
Via: SIP/2.0/UDP 172.16.0.50:5060;branch=z9hG4bK0b53edb0;rport•
Max-Forwards: 70•
From: “asterisk” sip:asterisk@172.16.0.50;tag=as5dbbcede•
To: sip:9002@172.16.0.100:5060;transport=udp
Contact: sip:asterisk@172.16.0.50:5060
Call-ID: 7d75f1fd39f2e8815a2772ef1219aa04@172.16.0.50:5060•
CSeq: 102 OPTIONS•
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3•
Date: Wed, 17 Jun 2015 16:02:52 GMT•
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH•
Supported: replaces, timer•
Content-Length: 0•

OPTIONS sip:9002@172.16.0.100:5060;transport=udp SIP/2.0•
Via: SIP/2.0/UDP 172.16.0.50:5060;branch=z9hG4bK0b53edb0;rport•
Max-Forwards: 70•
From: “asterisk” sip:asterisk@172.16.0.50;tag=as5dbbcede•
To: sip:9002@172.16.0.100:5060;transport=udp
Contact: sip:asterisk@172.16.0.50:5060
Call-ID: 7d75f1fd39f2e8815a2772ef1219aa04@172.16.0.50:5060•
CSeq: 102 OPTIONS•
User-Agent: Asterisk PBX 1.8.13.1~dfsg1-3+deb7u3•
Date: Wed, 17 Jun 2015 16:02:52 GMT•
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH•
Supported: replaces, timer•
Content-Length: 0•

REGISTER sip:172.16.0.50 SIP/2.0•
Via: SIP/2.0/UDP 172.16.0.100:5060;branch=z9hG4bK35f2319b•
From: sip:9002@172.16.0.50;tag=0018ba722bb90010524f6823-1681727b•
To: sip:9002@172.16.0.50
Call-ID: 0018ba72-2bb90002-e6b0a538-4ad2dd5c@172.16.0.100
Max-Forwards: 70•
Date: Wed, 17 Jun 2015 16:02:38 GMT•
CSeq: 115 REGISTER•
User-Agent: Cisco-CP7941G/8.0•
Contact: sip:9002@172.16.0.100:5060;transport=udp;+sip.instance="<urn:uuid:00000000-0000-000
0-0000-0018ba722bb9>";+u.sip!model.ccm.cisco.com=“115”•
Content-Length: 0•
Expires: 3600•[/quote]

Thanks in advance again…
I hope someone can help me