Asterisk phone Cisco SCCP 7970

Here you can find explanation how to setup Cisco 7970 IP Phone with SCCP image to work on Asterisk.



1. Download Asterisk chan_sccp-b from http://sourceforge.net/projects/chan-sccp-b/

2. edit /etc/asterisk/sccp.conf so it looks something like this:

[devices]

type        = 7970		; device type (see below)
autologin   = 30,31,		; lines list. You can add an empty line for an empty button (7960, 7970, 7940, 7920)
description = jj7970		; internal description. Not important
tzoffset  = -9
transfer = on			; enable or disable the transfer capability. It does remove the transfer softkey
park = on				; take a look to the compile howto. Park stuff is not compiled by default
speeddial =				; you can add an empty speedial if you want an empty button (7960, 7970, 7920)
speeddial = *97,voicemail,
cfwdall = off			; activate the callforward stuff and softkeys
cfwdbusy = off
dtmfmode = inband			; inband or outofband. outofband is the native cisco dtmf tone play.
					; Some phone model does not play dtmf tones while connected (bug?), so the default is inband
; imageversion = P00405000700	; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server)
; deny=0.0.0.0/0.0.0.0				; Same as general
; permit=10.0.0.0/255.255.255.0		; This device can register only using this ip address
permit=10.0.0.175 /255.255.255.255
dnd = on						; turn on the dnd softkey for this device. Valid values are "off", "on" 
							; (busy signal), "reject" (busy signal), "silent" (ringer = silent)
trustphoneip = no			; The phone has a ip address. It could be private, so if the phone is behind NAT
					; we don't have to trust the phone ip address, but the ip address of the connection
;earlyrtp = none			; valid options: none, offhook, dial, ringout. default is none.
					; The audio strem will be open in the progress and connected state.
private = on			; permit the private function softkey for this device
mwilamp = on			; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
mwioncall = off			; Set the MWI on call.
device => SEP0016C87754CE	; device name SEP<MAC>

[lines]

id          = 30			; future use
pin         = 1234		; future use
label       = 30			; button line label (7960, 7970, 7940, 7920)
description = Line 30		; top diplay description
context     = sip			; sccp
incominglimit = 2			; more than 1 incoming call = call waiting.
transfer = on			; per line transfer capability. on, off, 1, 0
mailbox = 30			; voicemail.conf (syntax: vmbox[@context][:folder])
vmnum = *97				; speeddial for voicemail administration, just a number to dial
cid_name = JJJ			; caller id name 
cid_num = 30
trnsfvm = 1000				; extension to redirect the caller (e.g for voicemail)
secondary_dialtone_digits = 9		; digits for the secondary dialtone (max 9 digits)
secondary_dialtone_tone = 0x21	; outside dialtone
music			; Sets the default music on hold class
language=en					; Default language setting
;accountcode=79501			; accountcode to ease billing
rtptos = 184				; sets the the rtp packets TOS for this line
echocancel = on				; sets the phone echocancel for this line
silencesuppression = off		; sets the silence suppression for this line
;callgroup=1,3-4				; We are in caller groups 1,3,4. Valid for this line
;pickupgroup=1,3-5			; We can do call pick-p for call group 1,3,4,5. Valid for this line
;amaflags =					; Sets the default AMA flag code stored in the CDR record for this line
line => 30

3. edit your /etc/asterisk/extensions.conf
To be added.

4. In root directory of your tftp server put this file SEP<MAC>.cnf.xml which looks like this:

<device  xsi:type="axl:XIPPhone">
<devicePool>
<name>Default</name>
<dateTimeSetting>
<name>CMLocal</name>
<dateTemplate>y-M-D</dateTemplate>
<timeZone>W. Europe Standard/Daylight Time</timeZone>
</dateTimeSetting>
<callManagerGroup>
<members>
<member  priority="0">
<callManager>
<ports>

<ethernetPhonePort>2000</ethernetPhonePort>
</ports>
<processNodeName>10.0.0.83</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo>
<name>Enable</name>
<srstOption>Enable</srstOption>
<userModifiable>true</userModifiable>

<ipAddr1>10.0.0.83</ipAddr1>
<port1>2000</port1>

<ipAddr2></ipAddr2>
<port2>2000</port2>

<ipAddr3></ipAddr3>
<port3>2000</port3>

</srstInfo>
<mlppDomainId>-1</mlppDomainId>
<mlppIndicationStatus>Default</mlppIndicationStatus>
<preemption>Default</preemption>
</devicePool>
<loadInformation></loadInformation>

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

<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>1</webAccess>
<daysDisplayNotActive>1,7</daysDisplayNotActive>
<displayOnTime>08:30</displayOnTime>
<displayOnDuration>11:30</displayOnDuration>
<displayIdleTimeout>01:00</displayIdleTimeout>
</vendorConfig>
<versionStamp></versionStamp>
<userLocale>
<name></name>
<uid>1</uid>

<langCode>en</langCode>
<version>4.0(1)</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale></networkLocale>
<networkLocaleInfo>
<name></name>
<uid>64</uid>
<version>4.0(1)</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>

<idleTimeout>120</idleTimeout>
<authenticationURL></authenticationURL>
<directoryURL>http://192.168.1.240/directory.php</directoryURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<servicesURL>http://192.168.1.240/services.php</servicesURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
</device>


Now, when phone boots up it will download SEP<MAC>.cnf.xml from tftp server. Then he will register with asterisk. And if you have setup extensions.conf corecty you can dial and receive calls.


Created by: parcina, Last modification: Tue 08 of May, 2012 (05:18 UTC) by admin
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