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.


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
Please update this page with new information, just login and click on the "Edit" or "Discussion" tab. Get a free login here: Register Thanks! - Find us on Google+