ATKEN Techniki | Cisco 79xx with Asterisk SCCP
VoIP PBXs, Datacenters, Cabling Installation, KNX
VoIP PBXs, Datacenters, Cabling Installation, KNX
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Cisco 79xx with Asterisk SCCP

Cisco 79xx with Asterisk SCCP

Which Cisco 79??. It could be a later model, which you could flash it so as to use SIP.

Anyway, try the below:

1. Download and install sccp, Chan_SCCP.

cd /usr/src

wget https://downloads.sourceforge.net/project/chan-sccp-b/V3/Chan_SCCP-3.0_RC3.tar.gz

tar xvf Chan_SCCP-3.0_RC3.tar.gz

cd Chan_SCCP-3.0_RC3

./Configure

make

make install

2.If there is no TFTP server on your pbx, please do install (it exists on elastix).

 

3.Edit and update the file /etc/asterisk/modules.conf:

noload => chan_skinny.so

load => chan_sccp.so
4.Disable sieve on ASTERISK, since it uses port 2000.

vi /etc/cyrus.conf

# UNIX sockets start with a slash and are put into /var/lib/imap/sockets

SERVICES {

# add or remove based on preferences

imap          cmd=»imapd» listen=»imap» prefork=5

imaps         cmd=»imapd -s» listen=»imaps» prefork=1

pop3          cmd=»pop3d» listen=»pop3″ prefork=3

pop3s         cmd=»pop3d -s» listen=»pop3s» prefork=1

#sieve         cmd=»timsieved» listen=»sieve» prefork=0
5.Create the files OS79XX.TXT, XMLDefault.cnf.xml and SEP<MAC>.cnf.xml (where <MAC> is the MAC address of the phone), using the below contents. The files must be in /tftpboot/ or inside the directory that TFT server uses.


OS79XX.TXT:

It contains the phone firmware. It is different for each model. The below is for 7910.

cd /tftpboot/

[root@pbx tftpboot]# cat OS79XX.TXT

P00405000700.bin #The files P00405000700.bin and P00405000700.sbn should be in /tftpboot/.

XMLDefault.cnf.xml:

[root@pbx tftpboot]# cat XMLDefault.cnf.xml

<Default>

<callManagerGroup>

<members>

<member priority=»0″>

<callManager>

<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

</ports>

<processNodeName>ASTERISK IP</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

<loadInformation6 model=»IP Phone 7910″>P00405000700</loadInformation6> #Change this, based on your Cisco model!

</Default>

SEP0004C1C69651.cnf.xml #Change this MAC 0004C1C69651 with the one located on the back of your phone.

[root@pbx tftpboot]# cat SEP0004C1C69651.cnf.xml

<device>

<devicePool>

<dateTimeSetting>

<dateTemplate>DMY</dateTemplate>

<timeZone>W. Europe Standard/Daylight Time</timeZone>

<ntps>

<ntp>

<name>193.92.150.3</name>

<ntpMode>Unicast</ntpMode>

</ntp>

</ntps>

</dateTimeSetting>

<callManagerGroup>

<members>

<member priority=»0″>

<callManager>

<ports>

<ethernetPhonePort>2000</ethernetPhonePort>

</ports>

<processNodeName>ASTERISK IP</processNodeName>

</callManager>

</member>

</members>

</callManagerGroup>

</devicePool>

<versionStamp>{Jan 01 2005 00:00:00}</versionStamp>

<loadInformation>P00405000700</loadInformation> #Change this, based on your Cisco model!

<userLocale>

<name>English_United_States</name>

<langCode>en</langCode>

</userLocale>

<networkLocale>United_States</networkLocale>

<idleTimeout>0</idleTimeout>

</device>

SCCP.conf:

[root@pbx asterisk]# cat /etc/asterisk/sccp.conf

;================================================================================

;

; general definitions

;

;================================================================================

[general]

servername = VOIPASK #Change it to anything you want.

keepalive = 30

debug = 1

context = from-internal

dateFormat = D.M.Y

bindaddr = ASTERISK IP

port = 2000

disallow=all

allow=alaw

allow=ulaw

allow=g729

firstdigittimeout = 13

digittimeout = 3

digittimeoutchar = #

autoanswer_ring_time = 1

autoanswer_tone = 0x32

remotehangup_tone = 0x32

transfer_tone = 0

callwaiting_tone = 0x2d

musicclass=default

language=en

deny=0.0.0.0/0.0.0.0

permit=192.168.100.0/255.255.255.0 #Your subnet.

dnd = on

sccp_tos = 0x68

sccp_cos = 4

audio_tos = 0xB8

audio_cos = 6

video_tos = 0x88

video_cos = 5

echocancel = on

silencesuppression = off

trustphoneip = no

private = on

callanswerorder=oldestfirst

protocol=11

hotline_enabled=yes

hotline_context=default

hotline_extension=111

;================================================================================

;

; actual definitions

;

;================================================================================

[SEP0004C1C69651] #Change this MAC 0004C1C69651 with the one located on the back of your phone.

description = VOIPASK EXT  #Extension description.

devicetype = 7910 #Phone Model.

park = off

button = line, 1

cfwdall = off

type = device

keepalive = 30

tzoffset = +2

transfer = on

park = on

cfwdall = off

cfwdbusy = off

cfwdnoanswer = off

pickupexten = off

pickupcontext = sccp

pickupmodeanswer = on

dtmfmode = inband

imageversion = P00405000700 #Phone’s Firmware.

deny=0.0.0.0/0.0.0.0

permit=192.168.100.0/255.255.255.255 #Your subnet.

dnd = on

trustphoneip = no

nat=off

directrtp=on

earlyrtp = none

private = on

mwilamp = on

mwioncall = off

setvar=testvar=value

cfwdall = on

[1] #Add here the subscriber that you want to register with this phone.

id = 1

type = line

pin = PASSWORD #Obsolete?

label = Phone 1

description = Line 1

mailbox = 10011

cid_name = VOIPASK #What to display when you call someone.

cid_num = 1

accountcode=79011

callgroup=1,3-4

pickupgroup=1,3-5

context = from-internal

incominglimit = 2

transfer = on

vmnum = *97

meetme = on

meetmeopts = qxd

meetmenum = 700

trnsfvm = 1000

secondary_dialtone_digits = 9

secondary_dialtone_tone = 0x22

musicclass=default

language=en

audio_tos = 0xB8

audio_cos = 6

video_tos = 0x88

video_cos = 5

echocancel = on

silencesuppression = off

;create a user defined softkeyset

;valid softkeys:

;redial, newcall, cfwdall, cfwdbusy, cfwdnoanswer, pickup, gpickup, conflist, dnd, hold, endcall, park, select

;idivert, resume, newcall, transfer, dirtrfr, answer, transvm, private, meetme, barge, cbarge, conf, back join

[softkeyset]

type=softkeyset

onhook      = redial,newcall,cfwdall,dnd

connected   = hold,endcall,park,select,cfwdall,cfwdbusy,idivert

onhold      = resume,newcall,endcall,transfer,confrn,select,dirtrfr,idivert

ringin      = answer,endcall,idivert

offhook      = redial,endcall,private,cfwdall,cfwdbusy,pickup,gpickup,meetme,barge

conntrans   = hold,endcall,transfer,confrn,park,select,dirtrfr,cfwdall,cfwdbusy

digitsfoll   = back,endcall

connconf   = hold,endcall,join

ringout      = endcall,transfer,cfwdall,idivert

offhookfeat   = redial,endcall

onhint      = pickup,barge

On your ASTERISK PBX create a new extension as OTHER (Custom Device)! On dial field add SCCP/XXX, where XXX is the User Extension. For the above example, it would be SCCP/1.

Restart ASTERISK & Cisco IP Phone!

Good luck, since it is a big procedure.

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