Cisco 7971G-GE SIP v8.0.3

Pretty simple to get the basics working. What is NOT working at the moment (and could be due to the v8.0.3 firmware, I really need to find/use v8.0.2 which is the stable release):

1) MWI
2) Correct reading of dialplan.xml
3) DTMF tones (rather than the fixed single tone for keypresses!)
4) Hold Music

7/7/06 UPDATE: V8.0.2SR1 fixes the MWI issue and a reboot fixes the Hold Music issue as well! So only, the DTMF tones & dialplan reading remain. Then on to the new toys of the phone such as the softkey's at the bottom & HINTing of the line appearances on the right!

7970/7971 Shortcuts
Unlock/lock phone: **#
Reboot phone: **#**

All suggestions, tips or contributions would be greatly appreciated!

To the point: What I have done to get a new 7971G-GE with SCCP firmware to boot, load SIP firmware, and connect successfully/correctly to Asterisk.

TFTPBOOT FILES
===========
/tftpboot/apps70.1-1-2-26.sbn
/tftpboot/cnu70.3-1-2-26.sbn
/tftpboot/cvm70sip.8-0-2-25.sbn
/tftpboot/dsp70.1-1-2-26.sbn
/tftpboot/jar70sip.8-0-2-25.sbn
/tftpboot/load119.txt
If you have a 7970 you need the matching load30006.txt
/tftpboot/SIP70.8-0-3S.loads
/tftpboot/term71.default.loads
/tftpboot/XMLDefault.cnf.xml

  Make sure you have an entry for your phone in here!
  e.g. 7971 ... <loadInformation119  model="Cisco 7971">SIP70.8-0-3S</loadInformation119>
  e.g. 7970 ... <loadInformation30006  model="Cisco 7970">SIP70.8-0-3S</loadInformation30006>

/tftpboot/Desktops/320x212x12/List.xml

  Thumbnail needs to be 25% of fullsize. Fullsize must be 320x212 and 12-bit (4096) color PNG.

<CiscoIPPhoneImageList>
	<ImageItem Image="thumbnail.png" URL="background.png"/> 
</CiscoIPPhoneImageList>

/tftpboot/distinctiveringlist.xml

<CiscoIPPhoneRingList>
	<Ring>
		<DisplayName>Classic</DisplayName>
		<FileName>ring_classic.pcm</FileName>
	</Ring> 
	<Ring>
		<DisplayName>Merlin</DisplayName>
		<FileName>ring_merlin.pcm</FileName>
	</Ring> 
</CiscoIPPhoneRingList>

/tftpboot/SEPxxxxxx.cnf.xml

<device xsi:type="axl:XIPPhone" ctiid="203849429" uuid="{96f8508b-10ef-f98c-d20d-0471777ec725}"> 
	<fullConfig>true</fullConfig> 
	<deviceProtocol>SIP</deviceProtocol> 
	<sshUserId>root</sshUserId> 
	<sshPassword>root</sshPassword> 
	<devicePool uuid="{a755aa55-089c-2b47-9603-c7d51b9ca4b5}"> 
		<dateTimeSetting uuid="{9ec4850a-7748-11d3-bdf0-00108302ead1}"> 
			<dateTemplate>M/D/Ya</dateTemplate> 
			<timeZone>Pacific Standard/Daylight Time</timeZone> 
			<ntps> 
				<ntp>
					<name>pbx.yoursite.com</name> 
					<ntpMode>Unicast</ntpMode> 
				</ntp>
			</ntps>
		</dateTimeSetting> 
		<callManagerGroup>
			<tftpDefault>true</tftpDefault> 
			<members> 
				<member priority="0"> 
					<callManager> 
						<name>pbx.yoursite.com</name> 
						<description>Asterisk VoIP/PBX</description> 
						<ports>
							<ethernetPhonePort>2000</ethernetPhonePort> 
							<sipPort>5060</sipPort> 
							<securedSipPort>5061</securedSipPort> 
							<mgcpPorts> 
								<listen>2427</listen> 
								<keepAlive>2428</keepAlive> 
							</mgcpPorts> 
						</ports> 
						<processNodeName>pbx.yoursite.com</processNodeName> 
					</callManager> 
				</member> 
			</members> 
		</callManagerGroup> 
		<srstInfo uuid="{cd241e11-4a58-4d3d-9661-f06c912a18a3}"> 
			<name>Disable</name> 
			<srstOption>Disable</srstOption> 
			<userModifiable>false</userModifiable>
			<ipAddr1>pbx.yoursite.com</ipAddr1>
			<port1>2000</port1> 
			<ipAddr2></ipAddr2> 
			<port2>2000</port2> 
			<ipAddr3></ipAddr3> 
			<port3>2000</port3> 
			<sipIpAddr1>pbx.yoursite.com</sipIpAddr1> 
			<sipPort1>5060</sipPort1> 
			<sipIpAddr2></sipIpAddr2> 
			<sipPort2>5060</sipPort2> 
			<sipIpAddr3></sipIpAddr3> 
			<sipPort3>5060</sipPort3> 
			<isSecure>false</isSecure> 
		</srstInfo> 
		<mlppDomainId>-1</mlppDomainId> 
		<mlppIndicationStatus>Default</mlppIndicationStatus> 
		<preemption>Default</preemption> 
		<connectionMonitorDuration>120</connectionMonitorDuration> 
	</devicePool> 
	<sipProfile> 
		<sipProxies> 
			<backupProxy>pbx.yoursite.com</backupProxy> 
			<backupProxyPort>5060</backupProxyPort> 
			<emergencyProxy>pbx.yoursite.com</emergencyProxy> 
			<emergencyProxyPort>5060</emergencyProxyPort> 
			<outboundProxy>pbx.yoursite.com</outboundProxy> 
			<outboundProxyPort>5060</outboundProxyPort> 
			<registerWithProxy>true</registerWithProxy> 
		</sipProxies> 
		<sipCallFeatures> 
			<cnfJoinEnabled>true</cnfJoinEnabled> 
			<callForwardURI>x-cisco-serviceuri-cfwdall</callForwardURI> 
			<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI> 
			<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI> 
			<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI> 
			<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI> 
			<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI> 
			<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>none</preferredCodec> 
		<dtmfAvtPayload>101</dtmfAvtPayload> 
		<dtmfDbLevel>3</dtmfDbLevel> 
		<dtmfOutofBand>avt</dtmfOutofBand> 
		<alwaysUsePrimeLine>false</alwaysUsePrimeLine> 
		<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail> 
		<kpml>3</kpml> 
		<phoneLabel>A V Zehenni</phoneLabel> 
		<stutterMsgWaiting>2</stutterMsgWaiting> 
		<callStats>false</callStats> 
		<offhookToFirstDigitTimer>15000</offhookToFirstDigitTimer> 
		<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts> 
		<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig> 
		<startMediaPort>16384</startMediaPort> 
		<stopMediaPort>32766</stopMediaPort> 
		<sipLines> 
			<line button="1"> 
				<featureID>9</featureID> 
				<featureLabel>x123</featureLabel> 
				<proxy>pbx.yoursite.com</proxy> 
				<port>5060</port> 
				<name>123</name> 
				<displayName>John Doe</displayName> 
				<autoAnswer> 
					<autoAnswerEnabled>2</autoAnswerEnabled> 
				</autoAnswer> 
				<callWaiting>3</callWaiting> 
				<authName>123</authName> 
				<authPassword>myextpassword</authPassword> 
				<sharedLine>false</sharedLine> 
				<messageWaitingLampPolicy>3</messageWaitingLampPolicy> 
				<messagesNumber>*97</messagesNumber> 
				<ringSettingIdle>4</ringSettingIdle> 
				<ringSettingActive>5</ringSettingActive> 
				<contact>114</contact> 
				<forwardCallInfoDisplay> 
					<callerName>true</callerName> 
					<callerNumber>false</callerNumber> 
					<redirectedNumber>false</redirectedNumber> 
					<dialedNumber>true</dialedNumber> 
				</forwardCallInfoDisplay> 
			</line> 
			<line button="8"> 
				<featureID>2</featureID>
				<featureLabel>Pickup</featureLabel> 
				<speedDialNumber>*8</speedDialNumber> 
			</line>
		</sipLines>
		<voipControlPort>5060</voipControlPort> 
		<dscpForAudio>184</dscpForAudio> 
		<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy> 
		<dialTemplate>dialplan.xml</dialTemplate> 
		<softKeyFile>softkey.xml</softKeyFile> 
	</sipProfile> 
	<commonProfile> 
		<phonePassword></phonePassword> 
		<backgroundImageAccess>true</backgroundImageAccess> 
		<callLogBlfEnabled>2</callLogBlfEnabled> 
	</commonProfile> 
	<loadInformation>SIP70.8-0-3S</loadInformation> 
	<vendorConfig> 
		<disableSpeaker>false</disableSpeaker>
		<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
		<pcPort>0</pcPort>
		<settingsAccess>1</settingsAccess>
		<garp>0</garp>
		<voiceVlanAccess>0</voiceVlanAccess>
		<videoCapability>0</videoCapability>
		<autoSelectLineEnable>0</autoSelectLineEnable>
		<webAccess>1</webAccess>
		<daysDisplayNotActive>1,7</daysDisplayNotActive>
		<displayOnTime>08:00</displayOnTime>
		<displayOnDuration>12:00</displayOnDuration>
		<displayIdleTimeout>01:00</displayIdleTimeout>
		<spanToPCPort>1</spanToPCPort>
	</vendorConfig> 
	<versionStamp>1136931633-57191cee-5ffc-4342-b286-4246b4991890</versionStamp> 
	<userLocale> 
		<name>English_United_States</name> 
		<uid>1</uid> 
		<langCode>en_US</langCode> 
		<version>1.0.0.0-1</version> 
		<winCharSet>iso-8859-1</winCharSet> 
	</userLocale> 
	<networkLocale>United_States</networkLocale> 
	<networkLocaleInfo> 
		<name>United_States</name> 
		<uid>64</uid> 
		<version>1.0.0.0-1</version> 
	</networkLocaleInfo> 
	<deviceSecurityMode>1</deviceSecurityMode> 
	<idleTimeout>0</idleTimeout> 
	<idleURL></idleURL> 
	<authenticationURL>http://pbx.yoursite.com/cisco/services/7971-authenticate.php</authenticationURL> 
	<directoryURL>http://pbx.yoursite.com/cisco/services/PhoneDirectory.php</directoryURL> 
	<informationURL>http://pbx.yoursite.com/cisco/services/7971-help.php</informationURL> 
	<messagesURL></messagesURL> 
	<proxyServerURL></proxyServerURL> 
	<servicesURL>http://pbx.yoursite.com/cisco/services/index_cisco.php</servicesURL> 
	<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig> 
	<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices> 
	<dscpForCm2Dvce>96</dscpForCm2Dvce> 
	<transportLayerProtocol>4</transportLayerProtocol> 
	<capfAuthMode>0</capfAuthMode> 
	<capfList> 
		<capf> 
			<phonePort>3804</phonePort> 
			<processNodeName>ccm-beta-5-1</processNodeName> 
		</capf> 
	</capfList> 
	<certHash></certHash> 
	<encrConfig>false</encrConfig> 
	<natReceivedProcessing>true</natReceivedProcessing> 
	<natEnabled>false</natEnabled> 
	<natAddress></natAddress> 
</device>

Hello,

I've got a Cisco 7971G-GE under 8.0.3S (and a copy of 8.0.2SR1 that I haven't tried yet), and I was successful in getting the SIP firmware to load using your helpful example above.

However, I'm unable to get the phones to register with a sip username and password. As you don't mention this explicitly above, could you please explain this further, as the xml config is a little cryptic, and the Cisco docs appear to be less than helpful in this regard. Thanks for you help!

Is this the same config for the 7970?

My 7971G-GE won't access the directory (or services) URL. The phone just shows 'Requesting....' but never makes the request according to Apache log. My proxyServerURL tag is empty and the web server is on the local LAN. The directoryURL tag points to an IP address (the Asterisk box), so it shouldn't be a DNS issue. I can access the directory from a webserver on a PC.

Any thoughts?

I mirrored the phone's switch port to my PC running Ethereal and determined that the phone is not even sending out a request for the directory or services URL. If I make a phone call, Ethereal captures all sorts of packets. But absolutely nothing happens when I hit the Directory button on the phone. I've tried resetting to factory with no luck. I'll send you the config file by PM, but I don't think that is the problem.

I haven't been able to figure out how to debug the phone from SSH. The 'debug http' option of the 7960 phone does not exit on the 7971. Any ideas on how to do network debugging from the SSH prompt of the phone?

Yes, you're right. Will edit the original post! Thx.

When I click View Phone Directory (under the Earth Key - Services) on the 7970 I get the following error

XML Error [4]: Parse Error

Below is the config from the SEP[MAC].cnf.xml file

http://10.0.0.x/cisco/services/7971-authenticate.php
http://10.0.0.x/cisco/services/PhoneDirectory.php
http://10.0.0.x/cisco/services/7971-help.php

Any ideas why it is failing. It works fine on the 7960?

Thanks!

Thanks. Just wanted to make sure that I did not screw something up. I would offer to help but I have trouble spelling XML.

Does this explain anything?
http://weblog.barnet.com.au/edwin/000132.html

Guys

While I'm not using TrixBox I am trying to get a 7970 using 8.0.3 on a standard Asterisk System.

So far I've been able to get the phone upgraded and am able to dial the 7970 from the console successfully, however whenever I dial a number from the 7970 I get the FAST BUSY tone after the first digit.

In the SIP debug I have the following (note the first digit dialled is a 7)

<-- SIP read from 10.131.111.51:49226:
INVITE sip:7@10.131.111.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8
From: "301" ;tag=000f3487e566000ed76e9557-dcf4e93a
To:
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51
Max-Forwards: 70
Date: Thu, 20 Jul 2006 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7970G/8.0
Contact:
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Remote-Party-ID: "301"

;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Allow-Events: kpml,dialog
Content-Length: 275
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 21184 0 IN IP4 10.131.111.51
s=SIP Call
t=0 0
m=audio 19796 RTP/AVP 0 8 18 101
c=IN IP4 10.131.111.51
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (19 headers 13 lines)---
Using INVITE request as basis request - 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51
Sending to 10.131.111.51 : 5060 (non-NAT)
Found user '301'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.131.111.51:19796
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing),

combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1

(telephone-event)
Looking for 7 in 7970SIP (domain 10.131.111.10)
Reliably Transmitting (no NAT) to 10.131.111.51:5060:
SIP/2.0 484 Address Incomplete
Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8;received=10.131.111.51
From: "301" ;tag=000f3487e566000ed76e9557-dcf4e93a
To: ;tag=as17095de3
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact:
Content-Length: 0

---

<-- SIP read from 10.131.111.51:49227:
ACK sip:7@10.131.111.10;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.131.111.51:5060;branch=z9hG4bK8f6323d8
From: "301" ;tag=000f3487e566000ed76e9557-dcf4e93a
To: ;tag=as17095de3
Call-ID: 000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51
Date: Thu, 20 Jul 2006 GMT
CSeq: 101 ACK
Content-Length: 0

--- (8 headers 0 lines)---
Destroying call '000f3487-e5660009-e2e36f89-af6ced7c@10.131.111.51'

Any ideas please?

Wasn't a dialplan issue in the end......was a problem with username/auth (although the SIP debug wasn't picking it up).

We love working out Cisco file naming conventions!!!

Has anyone on this forum managed to get the MWI indicator on the phone working correctly?

That looks like the last thing I need to get functional before I go into tinkering mode.

Thanks KSDEV

I don't have 8.0.2 so I'll have to wait....when was 8.0.3 released and any idea when the next release is due?

You'd think Cisco could release a firmware that didn't break a pre-existing working feature....Ho Hum :lol:

As an aside, thanks to "zibi" here in the TB family I was able to cleanup/merge the config file I originally posted in this thread with a few mods found in his which seem to improve registration under the newer firmware. See below for a working copy (key values edited out of course!):

Search & Replace:
============
- "YOUR.SITE.HERE" with your TB domain or IP address
- "DISPLAY-TEXT" with your phone label for the upper right hand corner of the LCD
- "x123" with your LCD label for the extension you wish
- "123" with your extension
- "321" with your extension password

/tftpboot/SEPxxxxxx.cnf.xml

<device>
	<fullConfig>true</fullConfig>
	<deviceProtocol>SIP</deviceProtocol>
	<sshUserId>root</sshUserId>
	<sshPassword>root</sshPassword>
	<devicePool>
		<dateTimeSetting> 
			<dateTemplate>M/D/Ya</dateTemplate> 
			<timeZone>Pacific Standard/Daylight Time</timeZone> 
			<ntps> 
				<ntp>
					<name>YOUR.SITE.HERE</name> 
					<ntpMode>Unicast</ntpMode> 
				</ntp>
			</ntps>
		</dateTimeSetting>
		<callManagerGroup>
			<tftpDefault>true</tftpDefault>
			<members>
				<member priority="0">
					<callManager>
						<ports>
							<ethernetPhonePort>2000</ethernetPhonePort>
							<sipPort>5060</sipPort>
							<securedSipPort>5061</securedSipPort>
						</ports>
						<processNodeName>YOUR.SITE.HERE</processNodeName>
					</callManager>
				</member>
			</members>
		</callManagerGroup>
	</devicePool>
	<commonProfile>
		<phonePassword></phonePassword>
		<backgroundImageAccess>true</backgroundImageAccess>
		<callLogBlfEnabled>2</callLogBlfEnabled>
	</commonProfile>
	<loadInformation>SIP70.8-0-4SR1S</loadInformation>
	<vendorConfig>
		<disableSpeaker>false</disableSpeaker>
		<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
		<pcPort>0</pcPort>
		<settingsAccess>1</settingsAccess>
		<garp>0</garp>
		<voiceVlanAccess>0</voiceVlanAccess>
		<videoCapability>0</videoCapability>
		<autoSelectLineEnable>0</autoSelectLineEnable>
		<daysDisplayNotActive>1,7</daysDisplayNotActive>
		<displayOnTime>08:00</displayOnTime>
		<displayOnDuration>12:00</displayOnDuration>
		<displayIdleTimeout>0:10</displayIdleTimeout> 
		<webAccess>0</webAccess>
		<spanToPCPort>1</spanToPCPort>
		<loggingDisplay>1</loggingDisplay>
		<loadServer></loadServer>
	</vendorConfig>
	<userLocale>
		<name>English_United_States</name>
		<uid>1</uid>
		<langCode>en_US</langCode>
		<version>1.0.0.0-1</version>
		<winCharSet>iso-8859-1</winCharSet>
	</userLocale>
	<networkLocale>United_States</networkLocale> 
	<networkLocaleInfo> 
		<name>United_States</name> 
		<uid>64</uid> 
		<version>1.0.0.0-1</version> 
	</networkLocaleInfo> 
	<deviceSecurityMode>1</deviceSecurityMode>
	<authenticationURL>http://YOUR.SITE.HERE/cisco/services/authentication.php</authenticationURL>
	<directoryURL>http://YOUR.SITE.HERE/cisco/services/PhoneDirectory.php</directoryURL>
	<idleTimeout>0</idleTimeout>
	<idleURL></idleURL>
	<informationURL>http://YOUR.SITE.HERE/cisco/services/help.php</informationURL>
	<messagesURL></messagesURL>
	<proxyServerURL></proxyServerURL>
	<servicesURL>http://YOUR.SITE.HERE/cisco/services/index_cisco.php</servicesURL>
	<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
	<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
	<dscpForCm2Dvce>96</dscpForCm2Dvce>
	<transportLayerProtocol>4</transportLayerProtocol>
	<capfAuthMode>0</capfAuthMode>
	<capfList>
		<capf>
			<phonePort>3804</phonePort>
		</capf>
	</capfList>
	<certHash></certHash>
	<encrConfig>false</encrConfig>
	<sipProfile>
		<sipProxies>
			<backupProxy></backupProxy>
			<backupProxyPort></backupProxyPort>
			<emergencyProxy></emergencyProxy>
			<emergencyProxyPort></emergencyProxyPort>
			<outboundProxy></outboundProxy>
			<outboundProxyPort></outboundProxyPort>
			<registerWithProxy>true</registerWithProxy>
		</sipProxies>
		<sipCallFeatures>
			<cnfJoinEnabled>true</cnfJoinEnabled>
			<callForwardURI>x--serviceuri-cfwdall</callForwardURI>
			<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
			<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
			<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
			<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
			<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
			<rfc2543Hold>true</rfc2543Hold>
			<callHoldRingback>2</callHoldRingback>
			<localCfwdEnable>true</localCfwdEnable>
			<semiAttendedTransfer>true</semiAttendedTransfer>
			<anonymousCallBlock>2</anonymousCallBlock>
			<callerIdBlocking>2</callerIdBlocking>
			<dndControl>1</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>false</remotePartyID>
			<userInfo>None</userInfo>
		</sipStack>
		<autoAnswerTimer>1</autoAnswerTimer>
		<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
		<autoAnswerOverride>true</autoAnswerOverride>
		<transferOnhookEnabled>false</transferOnhookEnabled>
		<enableVad>true</enableVad>
		<preferredCodec>none</preferredCodec>
		<dtmfAvtPayload>101</dtmfAvtPayload>
		<dtmfDbLevel>3</dtmfDbLevel>
		<dtmfOutofBand>avt</dtmfOutofBand>
		<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
		<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
		<kpml>3</kpml>
		<natEnabled>0</natEnabled>
		<natAddress></natAddress>
		<stutterMsgWaiting>2</stutterMsgWaiting>
		<callStats>false</callStats>
		<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
		<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
		<startMediaPort>16384</startMediaPort>
		<stopMediaPort>32766</stopMediaPort>
		<voipControlPort>5060</voipControlPort>
		<dscpForAudio>184</dscpForAudio>
		<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
		<dialTemplate>dialplan.xml</dialTemplate>
		<phoneLabel>DISPLAY-TEXT</phoneLabel>
		<sipLines>
			<line button="1">
				<featureID>9</featureID>
				<featureLabel>x123</featureLabel>
				<name>123</name>
				<displayName>x123</displayName>
				<contact>123</contact>
				<proxy>YOUR.SITE.HERE</proxy>
				<port>5060</port>
				<autoAnswer>
					<autoAnswerEnabled>2</autoAnswerEnabled>
				</autoAnswer>
				<callWaiting>3</callWaiting>
				<authName>123</authName>
				<authPassword>321</authPassword>
				<sharedLine>false</sharedLine>
				<messageWaitingLampPolicy>3</messageWaitingLampPolicy>
				<messagesNumber>*97</messagesNumber>
				<ringSettingIdle>4</ringSettingIdle>
				<ringSettingActive>5</ringSettingActive>
				<forwardCallInfoDisplay>
					<callerName>true</callerName>
					<callerNumber>false</callerNumber>
					<redirectedNumber>false</redirectedNumber>
					<dialedNumber>true</dialedNumber>
				</forwardCallInfoDisplay>
			</line>
			<line button="8">
				<featureID>2</featureID>
				<featureLabel>Pickup</featureLabel>
				<speedDialNumber>*8</speedDialNumber>
			</line>
		</sipLines>
	</sipProfile>
</device>

NOTE: Here is a good simple North American dialplan for Cisco handsets that covers key features including instant dialing when transferring a call to voicemail for example ... also 7940/7960/7970/7971 friendly format!

/tftpboot/dialplan.xml

<DIALTEMPLATE>
	<TEMPLATE MATCH="0" Timeout="2"/> <!-- PSTN Operator --> 
	<TEMPLATE MATCH="011*" Timeout="6"/> <!-- International calls --> 
	<TEMPLATE MATCH="911" Timeout="0" Route="Emergency"/> <!-- 911 -->
	<TEMPLATE MATCH="w!" Timeout="1" Route="Emergency" Rewrite="911"/> <!-- 911 in alpha mode -->
	<TEMPLATE MATCH="1.........." Timeout="0"/> <!-- Long Distance --> 
	<TEMPLATE MATCH="......." Timeout="0"/> <!-- Local numbers --> 
	<TEMPLATE MATCH=".........." Timeout="6" Rewrite="1.........."/> <!-- CallerID autodial rewrite --> 
	<TEMPLATE MATCH="\*..." Timeout="0"/> <!-- Transfer to voicemail --> 
	<TEMPLATE MATCH="*" Timeout="15"/> <!-- Anything else --> 
</DIALTEMPLATE>

I think I am seeing an old bug returned yet again:

- MWI no longer working again, under v8.0.4SR1

I have tried the values 3 & 1 (according to docs at voip-info.org for v8.0.2SR1 3 is the ideal value) and get no MWI when voicemails are left either way.

Anyone else or could this be a 7971G firmware related quirk but not so much for 7970G users?

ksDevGuy

Sadly that is the truth. Cisco's commitment to SIP is better in their PR department than in engineering (I guess for obvious reasons, I don't begrudge them in general).

My v8.0.4 7971G hung & rebooted itself for no apparent reason the other day ... yet another fun anomaly of the 8.0.4 release I am assuming. Fun!

ksDevGuy

You may have already checked this but go to the extension's page in freePBX. Check to make sure that nat=never and qualify=no. For some reason, these are often set by default to nat=yes and qualify=yes. This will prevent the Ciscos from registering with Asterisk.

I do.. pm me.

-Percy