Phone
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Phone System
To-Do
- Get Callcentric DID (or Other)
- Edit Sip.conf and Extension.conf With Callcentric login and google account
- Test (test the DTMF TONES, calling, other)
- Change Voicemail Settings (send wav to someone)
- Add A Custom Voicemail Greating
- Add/Change Phone book (xml) (started )
- fix asterisk Config errors (some syntax changes needed in some files due to update)
- Make a web page for the phone system
- ADD URL (sip calling) calling into asterisk (google how to?)
Prerequisites
Supplies
- Sip Phone (cisco sip)
- Debian Server
- Callcentric DID
- Google Voice account
- Python 2.7 w/ setuptools and simplejson
- pygooglevoice script
- tftp (phone config)
- apache2 (host directory and logo)
What was installed
- python-setuptools
- pygooglevoice script
- tftp
- apache2
- asterisk 13
Installing Pygooglevoice
- We Ran the below code at root mainly to insure the install of pygooglevoice-0.5.tar.gz
wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz tar zxvf pygooglevoice-0.5.tar.gz cd pygooglevoice-0.5 python setup.py install sed -i 's|https://www.google.com/accounts/ServiceLoginAuth?service=grandcentral%7Chttps://accounts.google.com/ServiceLogin?service=grandcentral\&continue=https://www.google.com/voice%7C' /usr/local/lib/python2.7/dist-packages/googlevoice/settings.py
sed -i 's| galx.*| galx = re.search(r\"name=\\"GALX\\" type=\\"hidden\\"\\
*value=\\"(.+)\\"\", content).group(1)|' /usr/local/lib/python2.7/dist-packages/googlevoice/voice.py
- Latter We installed asterisk 13 from source and Apache+php using the Package manager as well as the tftp server we decided on using.
Main Files
TFTP
The Main files here are:
- dialplan.xml = dialplan on Phone ONLY
- P0S3-08-12-00.load = Phone Firmware
- XMLDefault.cnf.xml = All Cisco Phone config
- SIP(MAC_HERE).CNF = Phone config
- SipDEFAULT.CNF = All Cisco SIP Phones config
- *.PCM and *.RAW are ringtones (config-ed in RINGLIST.DAT)
HTTP
- NOTE: HTTP on cisco requires port usage in urls it will not assume port 80
- The Directory is located here! and the directory is a simple XML file (freeside.php with xml headers) ever there is a limit of 32 entries be warned If we need more we will need to use a database and fetch from there using a different script
- the Phone logo I also here
Asterisk Setup
We edited sip.conf and extensions.conf to our info. Adding a password to the phone and the correct Callcentric and Google info. Note: after editing the documents you must restart asterisk. After the asterisk server was up we set up the TFTP server with the config files for the phone and the ringtones.. the config file had to be added to to include the network IP of the TFTP server, the web-server and asterisk install and the creds. we gave this user in sip.conf
Files
Sip.conf
[general] context=phone ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) externrefresh=60 localnet=192.168.1.0/255.255.255.0 udpbindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls canreinvite=no dtmfmode = rfc2833 tcpenable=no ;directmedia=yes register => sipusername:sippassword@iptel.org:5060/253xxxxxxx session-timers=refuse disallow=all allow=ulaw allow=gsm [201] defaultuser=201 secret=secret1 type=peer callerid="user1 " host=dynamic context=phone outgoinglimit=1 incominglimit=1 canreinvite=no nat=yes qualify=yes [202] defaultuser=202 secret=secret2 type=peer callerid="user2 " host=dynamic context=phone outgoinglimit=1 incominglimit=1 canreinvite=no nat=yes qualify=yes [203] defaultuser=203 secret=secret3 type=peer callerid="user3 " host=dynamic context=phone outgoinglimit=1 incominglimit=1 canreinvite=no nat=yes qualify=yes [DID] context=sip defaultuser=sipusername type=peer secret=sippassword host=iptel.org fromdomain=iptel.org fromuser=253xxxxxxx trustrpid = yes sendrpid = yes canreinvite = no insecure=port,invite nat=yes
extensions.conf
[general]
static=no
writeprotect=no
autofallthrough=yes
clearglobalvars=yes
priorityjumping=no
[globals]
gtimeout=50 ; timeout value
; initialize
gvuser=10000
[sip]
exten => _253xxxxxxx,1,ExecIf($[${gvuser}!=10000]?Bridge(${gvuser}):Dial(SIP/201&SIP/203,60,D(:1)))
exten => _253xxxxxxx,n, Set(GLOBAL(gvuser)=10000)
exten => _253xxxxxxx, n, Hangup()
[phone]
include => sip
include => gv-outbound
[gv-outbound]
exten => _NXXNXXXXXX,1,GoTo(1${EXTEN},1)
exten => _1NXXNXXXXXX,1,Answer
exten => _1NXXNXXXXXX,n,Set(GLOBAL(gvuser)=${CHANNEL})
exten => _1NXXNXXXXXX,n,System(gvoice -e gvusername@gmail.com -p gvpassword call ${EXTEN} 1253xxxxxxx 1 &)
exten => _1NXXNXXXXXX,n,Ringing
exten => _1NXXNXXXXXX,n,Wait(30)
exten => _X.,n,Noop(Never received callback from Google Voice on channel ${gvuser} . exiting)
exten => h,1,GotoIf($["${CHANNEL(state)}" = "Ring"]?:bridged)
exten => h,n,Noop(Hangup on channel ${gvuser})
exten => h,n,System(gvoice -e gvusername@gmail.com -p gvpassword cancel &)
exten => h,n, Set(GLOBAL(gvuser)=10000)
exten => h,n,Hangup()
exten => h,n(bridged),Noop(The channel has been bridged successfully)
exten => h,n, Set(GLOBAL(gvuser)=10000)
Test cmds
gvoice -e [gvusername@gmail.com] -p [gvpassword] call NXXNXXXXXX [callcentricdid] 1
- the above makes a call
asterisk -rvvvvv
- (loads the asterisk server Diagnostic program, type help for list of cmds) some are: sip show peers sip show registry sip show users sip reload