Cisco 7970
- From: jason justman <jason (at) jasonjustman.com>
- Date: Fri, 24 Mar 2006 09:34:17 -0800
Best bet is to get "Asterisk Chan_Sccp" http://chan-sccp.berlios.de/
1.) setup your /etc/asterisk/sccp.conf with something like:
[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=192.168.1.90/255.255.255.255 ; This device can register only
using this ip address
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 => SEP00131A1F6366 ; 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 = from-internal ; 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[ (at) 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
musicclass=default ; 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
(do the same for line 31)
2.) setup lines 30/31 as a custom extension in astersik (i used amp)
and had it dial SCCP/30 and SCCP/31 as needed
3.) setup /tftpboot config for SEP<MAC>.xml
<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>(ASTERISK IP HERE)</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
<srstInfo>
<name>Enable</name>
<srstOption>Enable</srstOption>
<userModifiable>true</userModifiable>
<ipAddr1>(ASTERISK IP HERE)</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>
Tomislav Parčina wrote:
I have search wiki, asteriskguru, chan_sccp and some other site's for information's how to upgrade, and make Cisco 7970 IP phone to work with asterisk on SCCP firmware.
I'm sure that there are users on this group that have working Cisco 7970 phone. Please send me some information's how to do that.
--
Tomislav Parcina
tparcina#lama.hr
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- From: Tomislav Parčina <tparcina (at) lama.hr>