7970 Registration?

Asterisk Asterisk Support
1

I upgraded my 7970 to image: SIP70.8-2-2SR1 without problems.

I set the proper config ( I hope ) but the phone won’t register. It’s stuck in “Registering”. I am using Asterisk 1.2.13 but also tried a 1.4 version with the same result. In the Asterisk SIP Debug I see this:

<— SIP read from 192.168.0.233:49170 —>
REGISTER sip:10.1.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKa1315e8c
From: sip:c11@10.1.0.1;tag=0017956dcf220002e862c608-e366191c
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-df5682d8-9da5af2c@192.168.0.233
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:19:21 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: sip:c11@192.168.0.233:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0017956dcf22”;+u.sip!model.ccm.cisco.com="30006"
Content-Length: 0
Expires: 3600

<------------->
— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.233 : 49170 (NAT)

<— Transmitting (no NAT) to 192.168.0.233:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKa1315e8c;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf220002e862c608-e366191c
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-df5682d8-9da5af2c@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:c11@10.1.0.1
Content-Length: 0

<------------>
managed5*CLI>
<— Transmitting (no NAT) to 192.168.0.233:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKa1315e8c;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf220002e862c608-e366191c
To: sip:c11@10.1.0.1;tag=as22f8f288
Call-ID: 0017956d-cf220002-df5682d8-9da5af2c@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: sip:c11@192.168.0.233:5060;transport=udp;expires=3600
Date: Tue, 24 Apr 2007 04:20:07 GMT
Content-Length: 0

On the phone, in the web console logs I get:

DBG 14:47:09.443201 JVM: platform_get_phrase_text: index=1058
DBG 14:47:09.443858 JVM: platform_get_phrase_text: got phrase=REG send failure: REGISTER
ERR 14:47:09.444214 JVM: %REG send failure: REGISTER
DBG 14:47:09.444533 JVM: LINE 51/1: Registration state change: SIP_REG_STATE_IDLE —> SIP_REG_STATE_REGISTERING

and in SSH a show register displays:

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: IDLE
line APR state timer expires proxy:port

1 .1x REGISTERING 0 0 10.1.0.1:5060
2 … NONE 0 0 undefined:0
3 … NONE 0 0 undefined:0
4 … NONE 0 0 undefined:0
5 … NONE 0 0 undefined:0
6 … NONE 0 0 undefined:0
7 … NONE 0 0 undefined:0
8 … NONE 0 0 undefined:0
1-BU .11 REGISTERED 3595 2901 10.1.0.1:5060

Note: APR is Authenticated, Provisioned, Registered

I know there have been lots of “Registering” problems and I want to know if they still exist and if there is a way fix them!

Thanks

2

[quote=“catharsis3k”]I upgraded my 7970 to image: SIP70.8-2-2SR1 without problems.

I set the proper config ( I hope ) but the phone won’t register. [/quote]

Can you show us the configuration on Asterisk, and the contents of the phone config file?

-jav

3

sure. I advanced a bit since last night.

The phone is still stuck in “Registering” mode and the conversation with Asterisk is:

<-- SIP read from 192.168.0.233:49156:
REGISTER sip:10.1.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bK709ab15a
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:16:06 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: sip:c11@192.168.0.233:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0017956dcf22”;+u.sip!model.ccm.cisco.com="30006"
Content-Length: 0
Expires: 3600

— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.233 : 5060 (NAT)
Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bK709ab15a;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:c11@10.1.0.1
Content-Length: 0

Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bK709ab15a;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1;tag=as5d24d2db
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="1c8f57a3"
Content-Length: 0

Scheduling destruction of call ‘0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233’ in 15000 ms
managed5*CLI>
<-- SIP read from 192.168.0.233:49156:
REGISTER sip:10.1.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKcbaae582
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
Max-Forwards: 70
Date: Thu, 29 Mar 2007 21:16:06 GMT
CSeq: 102 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: sip:c11@192.168.0.233:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0017956dcf22”;+u.sip!model.ccm.cisco.com="30006"
Authorization: Digest username=“c11”,realm=“asterisk”,uri=“sip:10.1.0.1”,response=“6b9f3e96218c08cd3fdf87bb923bcfdd”,nonce=“1c8f57a3”,algorithm=MD5
Content-Length: 0
Expires: 3600

— (13 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.233 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKcbaae582;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:c11@10.1.0.1
Content-Length: 0

Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKcbaae582;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200027c32cf78-b0026ce2
To: sip:c11@10.1.0.1;tag=as5d24d2db
Call-ID: 0017956d-cf220002-96acfc70-3b33a70a@192.168.0.233
CSeq: 102 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 3600
Contact: sip:c11@192.168.0.233:5060;transport=udp;expires=3600
Date: Tue, 24 Apr 2007 17:56:50 GMT
Content-Length: 0

However now on the phone the line 1-BU shows as fully registered!

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: IDLE
line APR state timer expires proxy:port

1 .1x REGISTERING 0 0 10.1.0.1:5060
2 … NONE 0 0 undefined:0
3 … NONE 0 0 undefined:0
4 … NONE 0 0 undefined:0
5 … NONE 0 0 undefined:0
6 … NONE 0 0 undefined:0
7 … NONE 0 0 undefined:0
8 … NONE 0 0 undefined:0
1-BU 111 REGISTERED 3595 2330 10.1.0.1:5060

Note: APR is Authenticated, Provisioned, Registered

the 111 means that 1-BU is: Authenticated, Provisioned and Registered. What is line 1-BU ???

Here are the configs:
Asterisk:

[c11]
type=friend
username=c11
secret=c11
context=from-sip
callerid=c11
host=dynamic
nat=no
canreinvite=yes
dtmfmode=auto
call-limit=5
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g729

SEP_MAC.cnf.xml

SIP user pass 
    <devicePool>
            <callManagerGroup>
                            <members>
                                            <member priority="0">
                                                            <callManager>
                                                                            <ports>
                                                                                            <ethernetPhonePort>2000</ethernetPhonePort>
                                                                                            <sipPort>5060</sipPort>
                                                                                            <securedSipPort>5061</securedSipPort>
                                                                            </ports>

                                                                            <processNodeName>10.1.0.1</processNodeName>

                                                            </callManager>
                                            </member>
                            </members>
            </callManagerGroup>

            <srstInfo>
                    <srstOption>Disable</srstOption>
                    <ipAddr1>10.1.0.1</ipAddr1>
                    <port1>2000</port1>
                    <ipAddr2></ipAddr2>
                    <port2>2000</port2>
                    <ipAddr3></ipAddr3>
                    <port3>2000</port3>
                    <sipIpAddr1>10.1.0.1</sipIpAddr1>
                    <sipPort1>5060</sipPort1>
                    <sipIpAddr2></sipIpAddr2>
                    <sipPort2>5060</sipPort2>
                    <sipIpAddr3></sipIpAddr3>
                    <sipPort3>5060</sipPort3>
                    <isSecure>false</isSecure>
            </srstInfo>

            <connectionMonitorDuration>120</connectionMonitorDuration>
    </devicePool>

    <sipProfile>

            <sipProxies>
                            <backupProxy>10.1.0.1</backupProxy>
                            <backupProxyPort>5060</backupProxyPort>
                            <emergencyProxy>10.1.0.1</emergencyProxy>
                            <emergencyProxyPort>5060</emergencyProxyPort>
                            <outboundProxy>10.1.0.1</outboundProxy>
                            <outboundProxyPort>5060</outboundProxyPort>
                            <registerWithProxy>true</registerWithProxy>
            </sipProxies>

            <sipCallFeatures>
                            <cnfJoinEnabled>true</cnfJoinEnabled>
                            <rfc2543Hold>true</rfc2543Hold>
                            <callHoldRingback>2</callHoldRingback>
                            <localCfwdEnable>true</localCfwdEnable>
                            <semiAttendedTransfer>true</semiAttendedTransfer>
                            <anonymousCallBlock>2</anonymousCallBlock>
                            <callerIdBlocking>2</callerIdBlocking>
                            <dndControl>0</dndControl>
                            <remoteCcEnable>true</remoteCcEnable>
            </sipCallFeatures>

            <sipStack>
                    <sipInviteRetx>6</sipInviteRetx>
                    <sipRetx>10</sipRetx>
                    <timerInviteExpires>180</timerInviteExpires>
                    <timerRegisterExpires>3600</timerRegisterExpires>
                    <timerRegisterDelta>5</timerRegisterDelta>
                    <timerKeepAliveExpires>120</timerKeepAliveExpires>
                    <timerSubscribeExpires>120</timerSubscribeExpires>
                    <timerSubscribeDelta>5</timerSubscribeDelta>
                    <timerT1>500</timerT1>
                    <timerT2>4000</timerT2>
                    <maxRedirects>70</maxRedirects>
                    <remotePartyID>true</remotePartyID>
                    <userInfo>None</userInfo>
            </sipStack>

            <autoAnswerTimer>1</autoAnswerTimer>
            <autoAnswerAltBehavior>false</autoAnswerAltBehavior>
            <autoAnswerOverride>true</autoAnswerOverride>
            <transferOnhookEnabled>false</transferOnhookEnabled>
            <enableVad>false</enableVad>
            <preferredCodec>g729a</preferredCodec>
            <dtmfAvtPayload>101</dtmfAvtPayload>
            <dtmfDbLevel>3</dtmfDbLevel>
            <dtmfOutofBand>avt</dtmfOutofBand>
            <alwaysUsePrimeLine>false</alwaysUsePrimeLine>
            <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
            <kpml>3</kpml>
            <phoneLabel>G-Tech</phoneLabel>
            <stutterMsgWaiting>2</stutterMsgWaiting>
            <callStats>false</callStats>
            <offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer>
            <silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
            <disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
            <startMediaPort>16384</startMediaPort>
            <stopMediaPort>32766</stopMediaPort>

            <sipLines>
                    <line button="1">
                                    <featureID>9</featureID>
                                    <featureLabel>Line 1</featureLabel>
                                    <proxy>10.1.0.1</proxy>
                                    <port>5060</port>
                                    <displayName>c11</displayName>
                                    <autoAnswer>
                                                    <autoAnswerEnabled>2</autoAnswerEnabled>
                                    </autoAnswer>
                                    <callWaiting>3</callWaiting>
                                    <authName>c11</authName>
                                    <authPassword>c11</authPassword>
                                    <sharedLine>false</sharedLine>
                                    <messageWaitingLampPolicy>3</messageWaitingLampPolicy>
                                    <messagesNumber>8500</messagesNumber>
                                    <ringSettingIdle>4</ringSettingIdle>
                                    <ringSettingActive>5</ringSettingActive>
                                    <name>c11</name>
                                    <forwardCallInfoDisplay>
                                                    <callerName>true</callerName>
                                                    <callerNumber>false</callerNumber>
                                                    <redirectedNumber>false</redirectedNumber>
                                                    <dialedNumber>true</dialedNumber>
                                    </forwardCallInfoDisplay>
                    </line>
            </sipLines>

    </sipProfile>

    <commonProfile>
                    <phonePassword></phonePassword>
                    <backgroundImageAccess>true</backgroundImageAccess>
                    <callLogBlfEnabled>2</callLogBlfEnabled>
    </commonProfile>

    <loadInformation>SIP70.8-2-2SR1S</loadInformation>

    <vendorConfig>
                    <disableSpeaker>false</disableSpeaker>
                    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
                    <pcPort>1</pcPort>
                    <settingsAccess>1</settingsAccess>
                    <garp>0</garp>

                    <voiceVlanAccess>0</voiceVlanAccess>
                    <videoCapability>0</videoCapability>
                    <autoSelectLineEnable>0</autoSelectLineEnable>
                    <webAccess>0</webAccess>
                    <daysDisplayNotActive>1,7</daysDisplayNotActive>
                    <displayOnTime>08:00</displayOnTime>
                    <displayOnDuration>10:30</displayOnDuration>
                    <displayIdleTimeout>01:00</displayIdleTimeout>
                    <spanToPCPort>1</spanToPCPort>
    </vendorConfig>

    <transportLayerProtocol>4</transportLayerProtocol>

    <capfAuthMode>0</capfAuthMode>
    <capfList>
            <capf>
                    <phonePort>3804</phonePort>
            </capf>
    </capfList>

    <certHash></certHash>
    <encrConfig>false</encrConfig>
4

I tried another change in my sip.conf.

host=192.168.0.233 instead of dynamic
canreinvite=no

I now get the following errors when sip debug enabled:

Apr 24 14:57:39 ERROR[36626]: chan_sip.c:6635 register_verify: Peer ‘c11’ is trying to register, but not configured as host=dynamic

Apr 24 14:57:47 NOTICE[36626]: chan_sip.c:11188 handle_request_register: Registration from ‘sip:c11@10.1.0.1’ failed for ‘192.168.0.233’ - Username/auth name mismatch

I am pretty sure the username/auth name are the same so what’s up

5

[georgec11]
type=friend
username=georgec11
secret=georgec11

Change “secret” for “password” and put the host back to dynamic, reload SIP and see what happens.

-jav

6

done. The registration conversation is now:

<-- SIP read from 192.168.0.233:49230:
REGISTER sip:10.1.0.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKedc09229
From: sip:c11@10.1.0.1;tag=0017956dcf2200021e4f386c-e058a6b3
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-c24a228e-f7b48c1d@192.168.0.233
Max-Forwards: 70
Date: Thu, 29 Mar 2007 22:42:12 GMT
CSeq: 101 REGISTER
User-Agent: Cisco-CP7970G/8.0
Contact: sip:c11@192.168.0.233:5060;transport=udp;+sip.instance=“urn:uuid:00000000-0000-0000-0000-0017956dcf22”;+u.sip!model.ccm.cisco.com="30006"
Content-Length: 0
Expires: 3600

— (12 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.233 : 5060 (NAT)
Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKedc09229;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200021e4f386c-e058a6b3
To: sip:c11@10.1.0.1
Call-ID: 0017956d-cf220002-c24a228e-f7b48c1d@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:c11@10.1.0.1
Content-Length: 0

Transmitting (no NAT) to 192.168.0.233:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.233:5060;branch=z9hG4bKedc09229;received=192.168.0.233
From: sip:c11@10.1.0.1;tag=0017956dcf2200021e4f386c-e058a6b3
To: sip:c11@10.1.0.1;tag=as649d16c7
Call-ID: 0017956d-cf220002-c24a228e-f7b48c1d@192.168.0.233
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 3600
Contact: sip:c11@192.168.0.233:5060;transport=udp;expires=3600
Date: Tue, 24 Apr 2007 19:22:57 GMT
Content-Length: 0

Phone debug prints:
ERR 22:42:12.234525 JVM: %REG send failure: REGISTER
ERR 22:42:38.910855 NTP: Thu Mar 29 22:42:38 2007
ERR 22:43:12.239633 JVM: %REG send failure: REGISTER

7

Well, so, the phone has this:

10.1.0.1

But it appears to be at some other address (192.168.0.233). I don’t know anything about your network, but your Asterisk server is getting the registration from that address, and sending it back to 10.1.0.1.

You will want to work on network settings…

-jav

8

that is actually something extra that doesn’t get used by the phone.

the phone config ( show config-cache on phone ) is:

startMediaPort : 16384
endMediaPort : 32766
callerIdBlocking : 0
anonymousCallBlock : 0
dndControl : 0
preferredCodec : g729a
dtmfOutofBand : avt
dtmfAvtPayload : 101
dtmfDbLevel : 3
line1_featureID : 9
line2_featureID : 255
line3_featureID : 255
line4_featureID : 255
line5_featureID : 255
line6_featureID : 255
line7_featureID : 255
line8_featureID : 255
line1_name : c11
line2_name :
line3_name :
line4_name :
line5_name :
line6_name :
line7_name :
line8_name :
line1_authName : c11
line2_authName :
line3_authName :
line4_authName :
line5_authName :
line6_authName :
line7_authName :
line8_authName :
line1_authPassword : **********
line2_authPassword : **********
line3_authPassword : **********
line4_authPassword : **********
line5_authPassword : **********
line6_authPassword : **********
line7_authPassword : **********
line8_authPassword : **********
line1_displayName : c11
line2_displayName :
line3_displayName :
line4_displayName :
line5_displayName :
line6_displayName :
line7_displayName :
line8_displayName :
line1_contact :
line2_contact :
line3_contact :
line4_contact :
line5_contact :
line6_contact :
line7_contact :
line8_contact :
proxy1_address : 10.1.0.1
proxy2_address :
proxy3_address :
proxy4_address :
proxy5_address :
proxy6_address :
proxy7_address :
proxy8_address :
proxy1_port : 5060
proxy2_port : 5060
proxy3_port : 5060
proxy4_port : 5060
proxy5_port : 5060
proxy6_port : 5060
proxy7_port : 5060
proxy8_port : 5060
line1_autoAnswerEnabled : 0
line2_autoAnswerEnabled : 0
line3_autoAnswerEnabled : 0
line4_autoAnswerEnabled : 0
line5_autoAnswerEnabled : 0
line6_autoAnswerEnabled : 0
line7_autoAnswerEnabled : 0
line8_autoAnswerEnabled : 0
line1_autoAnswerMode : speaker
line2_autoAnswerMode : auto answer with speakerphone
line3_autoAnswerMode : auto answer with speakerphone
line4_autoAnswerMode : auto answer with speakerphone
line5_autoAnswerMode : auto answer with speakerphone
line6_autoAnswerMode : auto answer with speakerphone
line7_autoAnswerMode : auto answer with speakerphone
line8_autoAnswerMode : auto answer with speakerphone
line1_callWaiting : 1
line2_callWaiting : 1
line3_callWaiting : 1
line4_callWaiting : 1
line5_callWaiting : 1
line6_callWaiting : 1
line7_callWaiting : 1
line8_callWaiting : 1
line1_sharedLine : 0
line2_sharedLine : 0
line3_sharedLine : 0
line4_sharedLine : 0
line5_sharedLine : 0
line6_sharedLine : 0
line7_sharedLine : 0
line8_sharedLine : 0
line1_msgWaitingLampPolicy : 3
line2_msgWaitingLampPolicy : 0
line3_msgWaitingLampPolicy : 0
line4_msgWaitingLampPolicy : 0
line5_msgWaitingLampPolicy : 0
line6_msgWaitingLampPolicy : 0
line7_msgWaitingLampPolicy : 0
line8_msgWaitingLampPolicy : 0
line1_ringSettingIdle : 4
line2_ringSettingIdle : 4
line3_ringSettingIdle : 4
line4_ringSettingIdle : 4
line5_ringSettingIdle : 4
line6_ringSettingIdle : 4
line7_ringSettingIdle : 4
line8_ringSettingIdle : 4
line1_ringSettingActive : 5
line2_ringSettingActive : 5
line3_ringSettingActive : 5
line4_ringSettingActive : 5
line5_ringSettingActive : 5
line6_ringSettingActive : 5
line7_ringSettingActive : 5
line8_ringSettingActive : 5
sipRetx : 10
sipInviteRetx : 6
timerT1 : 500
timerT2 : 4000
timerInviteExpires : 180
timerRegisterExpires : 3600
registerWithProxy : 1
backupProxy : 10.1.0.1
backupProxyPort : 5060
emergencyProxy : 10.1.0.1
emergencyProxyPort : 5060
outboundProxy : 10.1.0.1
outboundProxyPort : 5060
natReceivedProcessing : 0
userInfo : none
cnfJoinEnabled : 1
remotePartyID : 1
semiAttendedTransfer : 1
callHoldRingback : 0
stutterMsgWaiting : 0
cfwd_URL :
callStats : 0
autoAnswer : 0
localCfwdEnable : 1
timerRegisterDelta : 5
MaxRedirects : 70
rfc2543Hold : 1
ccm1_address : 10.1.0.1
ccm2_address :
ccm3_address :
ccm1_sipPort : 0
ccm2_sipPort : 0
ccm3_sipPort : 5060
ccm1_securedSipPort : 5060
ccm2_securedSipPort : 5060
ccm3_securedSipPort : 0
ccm1_securityLevel : 0
ccm2_securityLevel : 0
ccm3_securityLevel : 0
ccm1_isValid : 0
ccm2_isValid : 0
ccm3_isValid : 0
ccmTftp_ipAddr :
ccmTftp_port : 0
ccmTftp_securedSipPort : 0
ccmTftp_isValid : 0
ccmTftp_securityLevel : 0
ccmSrst_sipIpAddr1 :
ccmSrst_sipPort1 : 5060
ccmSrst_securedSipPort : 0
ccmSrst_isValid : 0
ccmSrst_securityLevel : 0
connectionMonitorDuration : 120
callPickupURI :
callPickupListURI :
callPickupGroupURI :
meetMeServiceURI :
callForwardUri :
abbreviatedDialURI :
callLogBlfEnabled : 0
remoteCcEnabled : 1
timerKeepaliveExpires : 120
timerSubscribeExpires : 120
timerSubscribeDelta : 5
transportLayerProtocol : 4
kpml : 3
natEnabled : 0
natAddress :
voipControlPort : 5060
myIpAddr : 192.168.0.233
myMacAddr : 0017:956d:cf22
enableVad : 0
autoAnswerAltBehavior : 0
autoAnswerTimer : 1
autoAnswerOverride : 1
offhookToFirstDigitTimer : 15000
silentPeriodBetweenCallWaitingBursts : 10
ringSettingBusyStationPolicy : 0
dscpForCm2Dvce : 96
speakerEnabled : 1
transferOnhookEnabled : 0
retainForwardInformation : 0
line1_index : 0
line2_index : 0
line3_index : 0
line4_index : 0
line5_index : 0
line6_index : 0
line7_index : 0
line8_index : 0

Furthermore, normal registration packets from another phone ( a 7960 ) look pretty much the same
EX:

— (11 headers 0 lines) —
Using latest REGISTER request as basis request
Sending to 192.168.0.95 : 5060 (NAT)
Transmitting (no NAT) to 192.168.0.95:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.95:5060;branch=z9hG4bK67f6e7b5;received=192.168.0.95
From: sip:c1@10.1.0.1;tag=000ab8f37ff5000231186814-12001099
To: sip:c1@10.1.0.1
Call-ID: 000ab8f3-7ff50002-6d6d0118-3dc22e88@192.168.0.95
CSeq: 101 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:c1@10.1.0.1
Content-Length: 0

9

I got this solved but the solution makes no sense!

Removing the following from the phone xml config did the trick:

                                   <forwardCallInfoDisplay>
                                                    <callerName>true</callerName>
                                                    <callerNumber>false</callerNumber>
                                                    <redirectedNumber>false</redirectedNumber>
                                                    <dialedNumber>true</dialedNumber>
                                    </forwardCallInfoDisplay>

anyone knows why ?