Difference between revisions of "Phone"

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(Created page with "= 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...")
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Currently the phone system accepts incoming calls and make calls [[Equipment/Cisco_7960 | Cisco 7960]] see Eldon or [[Matt|User:Msg4real]] for Help<br>
= Phone System =
The system has a Voicemail setup and -mails the BOD<br>
 
Currently takes calls using the XMPP PROTOCOL and/or SIP
= Phone System ON PI=
 
 
== To-Do ==
== 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)
* Change Voicemail Settings (send wav to someone)
* Create IVR 
* Add A Custom Voicemail Greating
* Add A Custom Voicemail Greating
* Add/Change Phone book (xml) (dir.php) (started )
* Add/Change Phone book (xml) (started )
* Mount Phones into Workshop
* fix asterisk Config errors (some syntax changes needed in some files due to update)
* Buy a 48v ~1.5 amp Power brick or 15.4W per phone (currently POE with Original brick) [add poe to phones]
* Make a web page for the phone system
* Make E911 NOTICE for phone (and list of 911 numbers by phone)
* ADD URL (sip calling) calling into asterisk (google how to?)
 
== Prerequisites ==       
=== Supplies ===
* Sip Phone (cisco sip)
* Callcentric DID
* Google Voice account
* RasPI 2 w/ RASPBX
* tftp (phone config)
   
=== What was installed ===
* OSS module
 
=== Configure the GUI ===
* setup extensions
* add trunks
* incoming routes
* add outgoing routes
* OSS endpoint manager allows for easier phone setup
 
=== Make files for phones ===
* Place a logo on a web server as defined in the phone service man
* Make directory (see links)  
 
 
 
= old way = 
It uses the Google Voice Bounce Back method Commonly used by others before XMPP was created. <strike>I have however after some careful research and configuring we should have Several XMPP Lines. However, there have been reports that google is turning XMPP off, They have or some other crap. It will happen but what most people are refering to is Google has been Changing the Login Security, and detects what we are doing as an insecure app in logging in. Thus we must go into setting and change to allow and someone must logon to the account once day to once every 2 months...
 
I have added XMPP calling on lines 3-5 and Line 2 is not configured correctly at this time stay tuned...  </strike>


== Prerequisites ==         
== Prerequisites ==         
=== Supplies ===
=== Supplies ===
* Sip Phone (cisco sip)
* Sip Phone (cisco [[sip]])
* Debian Server
* Debian Server
* Callcentric DID
* Callcentric DID
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* apache2
* apache2
* asterisk 13
* asterisk 13
 
     
 
     
==== Installing  Pygooglevoice ====
==== Installing  Pygooglevoice ====
* We Ran the below code at root mainly to insure the install of pygooglevoice-0.5.tar.gz
* 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
  wget http://pygooglevoice.googlecode.com/files/pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5.tar.gz
tar zxvf pygooglevoice-0.5.tar.gz
cd pygooglevoice-0.5
cd pygooglevoice-0.5
python setup.py install
python setup.py install
sed -i 's|https://www.google.com/accounts/ServiceLoginAuth?service=grandcentral|https://accounts.google.com/ServiceLogin?service=grandcentral\&continue=https://www.google.com/voice|' /usr/local/lib/python2.7/dist-packages/googlevoice/settings.py
sed -i 's|https://www.google.com/accounts/ServiceLoginAuth?service=grandcentral|https://accounts.google.com/ServiceLogin?service=grandcentral\&continue=https://www.google.com/voice|' /usr/local/lib/python2.7/dist-packages/googlevoice/settings.py
sed -i 's|      galx.*|      galx = re.search(r\"name=\\"GALX\\" type=\\"hidden\\"\\
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  
  *value=\\"(.+)\\"\", content).group(1)|' /usr/local/lib/python2.7/dist-packages/googlevoice/voice.py  


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=== Asterisk Setup ===
=== 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           
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 ====
==== Files ====
===== Sip.conf =====   
===== Sip.conf =====   
[notice]
<code>[general]</code><code>context=phone                ; Default context </code><code>for</code> <code>incoming calls</code><code>allowoverlap=no                ; Disable overlap dialing support. (Default is </code><code>yes</code><code>)</code><code>bindport=5060                  ; UDP Port to bind to (SIP standard port is 5060)</code><code>externrefresh=60</code><code>localnet=192.168.1.0</code><code>/255</code><code>.255.255.0</code><code>udpbindaddr=0.0.0.0            ; IP address to bind to (0.0.0.0 binds to all)</code><code>srvlookup=</code><code>yes</code>                   <code>; Enable DNS SRV lookups on outbound calls</code><code>canreinvite=no</code><code>dtmfmode = rfc2833</code><code>tcpenable=no</code><code>;directmedia=</code><code>yes</code><code>register =&gt; sipusername:sippassword@iptel.org:5060</code><code>/253xxxxxxx</code><code>session-timers=refuse</code><code>disallow=all</code><code>allow=ulaw</code><code>allow=gsm</code> <code>[201]</code><code>defaultuser=201</code><code>secret=secret1</code><code>type</code><code>=peer</code><code>callerid=</code><code>"user1 "</code><code>host=dynamic</code><code>context=phone</code><code>outgoinglimit=1</code><code>incominglimit=1</code><code>canreinvite=no</code><code>nat=</code><code>yes</code><code>qualify=</code><code>yes</code> <code>[202]</code><code>defaultuser=202</code><code>secret=secret2</code><code>type</code><code>=peer</code><code>callerid=</code><code>"user2 "</code><code>host=dynamic</code><code>context=phone</code><code>outgoinglimit=1</code><code>incominglimit=1</code><code>canreinvite=no</code><code>nat=</code><code>yes</code><code>qualify=</code><code>yes</code> <code>[203]</code><code>defaultuser=203</code><code>secret=secret3</code><code>type</code><code>=peer</code><code>callerid=</code><code>"user3 "</code><code>host=dynamic</code><code>context=phone</code><code>outgoinglimit=1</code><code>incominglimit=1</code><code>canreinvite=no</code><code>nat=</code><code>yes</code><code>qualify=</code><code>yes</code> <code>[DID]</code><code>context=sip</code><code>defaultuser=sipusername</code><code>type</code><code>=peer</code><code>secret=sippassword</code><code>host=iptel.org</code><code>fromdomain=iptel.org</code><code>fromuser=253xxxxxxx</code><code>trustrpid = </code><code>yes</code><code>sendrpid = </code><code>yes</code><code>canreinvite = no</code><code>insecure=port,invite</code><code>nat=</code><code>yes</code>
This is just where we started here for ref
[/notice]
[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 =====
===== extensions.conf =====
[notice]
<code>[general]</code><code>static=no</code><code>writeprotect=no</code><code>autofallthrough=</code><code>yes</code><code>clearglobalvars=</code><code>yes</code><code>priorityjumping=no</code> <code>[globals]</code><code>gtimeout=50    ; timeout value</code><code>; initialize</code><code>gvuser=10000</code> <code>[sip]</code><code>exten =&gt; _253xxxxxxx,1,ExecIf($[${gvuser}!=10000]?Bridge(${gvuser}):Dial(SIP</code><code>/201</code><code>&amp;SIP</code><code>/203</code><code>,60,D(:1)))</code><code>exten =&gt; _253xxxxxxx,n, Set(GLOBAL(gvuser)=10000)</code><code>exten =&gt; _253xxxxxxx, n, Hangup()</code><code>[phone]</code><code>include =&gt; sip</code><code>include =&gt; gv-outbound</code><code>[gv-outbound]</code><code>exten =&gt; _NXXNXXXXXX,1,GoTo(1${EXTEN},1)</code><code>exten =&gt; _1NXXNXXXXXX,1,Answer</code><code>exten =&gt; _1NXXNXXXXXX,n,Set(GLOBAL(gvuser)=${CHANNEL})</code><code>exten =&gt; _1NXXNXXXXXX,n,System(gvoice -e gvusername@gmail.com -p gvpassword call ${EXTEN} 1253xxxxxxx 1 &amp;)</code><code>exten =&gt; _1NXXNXXXXXX,n,Ringing</code><code>exten =&gt; _1NXXNXXXXXX,n,Wait(30)</code><code>exten =&gt; _X.,n,Noop(Never received callback from Google Voice on channel ${gvuser} . exiting)</code><code>exten =&gt; h,1,GotoIf($[</code><code>"${CHANNEL(state)}"</code> <code>= </code><code>"Ring"</code><code>]?:bridged)</code><code>exten =&gt; h,n,Noop(Hangup on channel ${gvuser})</code><code>exten =&gt; h,n,System(gvoice -e gvusername@gmail.com -p gvpassword cancel &amp;)</code><code>exten =&gt; h,n, Set(GLOBAL(gvuser)=10000)</code><code>exten =&gt; h,n,Hangup()</code><code>exten =&gt; h,n(bridged),Noop(The channel has been bridged successfully)</code><code>exten =&gt; h,n, Set(GLOBAL(gvuser)=10000)</code> 
This is just where we started here for ref
   
[/notice]
[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 =
= Test cmds =
   gvoice -e [gvusername@gmail.com] -p [gvpassword] call NXXNXXXXXX [callcentricdid] 1   
   gvoice -e [gvusername@gmail.com] -p [gvpassword] call NXXNXXXXXX [callcentricdid] 1   
* the above makes a call  
* the above makes a call  


asterisk -rvvvvv  
asterisk -rvvvvv (loads the asterisk server, type help for list of cmds) some are: sip show peers  sip show registry  sip show users  sip reload
* (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
           
           
= Links =
= Links =
== Cisco 79XX Phone ==
[http://www.cisco.com/c/dam/en/us/td/docs/voice_ip_comm/sip/proxies/2-1/white/paper/sipwp21.pdf Phone setup]<br>
[http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cuipph/7960g_7940g/sip/english/user/guide/user/sipuget.html#wp1013789 How to use the 7060g]<br>
[http://www.voip-info.org/wiki/view/Asterisk+Cisco+79XX+XML+Services Directory info] <BR>
[http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79xx cisco 79xx info]<br>
== Asterisk 13 Documentation ==
[https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Documentation Asterisk 13 Documentation]<br>
[https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial Application Dial]<br>
<br>
<br>
== Services ==
[https://www.gvsip.com/ GVsip] A Free to low cost gv sip gateway <br>
[http://www.callcentric.com/ callcentric] A DID and SIP provider <br>
[http://www.linphone.org/free-sip-service.html Linphone] A free DID to sip service (does not work well with gv due to abuse)<br>
==Other==
[http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/index.html Asterisk™: The Future of Telephony]<br>
<br>
[http://tech.iprock.com/?p=9784 BounceBack with Freepbx] &nbsp;&nbsp;&nbsp;[Guide]<br>
[http://nerdvittles.com/?p=12233 Incredible PBX for Asterisk-GUI (RasPi 2 Edition)] &nbsp;&nbsp;&nbsp;[Guide]<br>
[http://www.einhorn-net.de/headset/index_e.html Headset Jack Pinout] << Make a Intercom anyone?


=[[equipment/Cisco_7960| Cisco 7960]] POE=
[[Phone setup]]  
* The Phone uses the old POE STD and thus requires a POE CROSSOVER cable to swap pins 4+5 to 7+8
[[How to use the 7060g]]
* The Phone also can use 802.3af STD B (power on non data pairs)
[[DIrectory info]]
* For smart POEs put a 25k resistor across pins 1 and 3 and connect your phone to the switch using a crossover cable. (this will tell the POE to send power)

Revision as of 17:43, 10 March 2015

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 callsallowoverlap=no  ; Disable overlap dialing support. (Default is yes)bindport=5060  ; UDP Port to bind to (SIP standard port is 5060)externrefresh=60localnet=192.168.1.0/255.255.255.0udpbindaddr=0.0.0.0  ; IP address to bind to (0.0.0.0 binds to all)srvlookup=yes ; Enable DNS SRV lookups on outbound callscanreinvite=nodtmfmode = rfc2833tcpenable=no;directmedia=yesregister => sipusername:sippassword@iptel.org:5060/253xxxxxxxsession-timers=refusedisallow=allallow=ulawallow=gsm [201]defaultuser=201secret=secret1type=peercallerid="user1 "host=dynamiccontext=phoneoutgoinglimit=1incominglimit=1canreinvite=nonat=yesqualify=yes [202]defaultuser=202secret=secret2type=peercallerid="user2 "host=dynamiccontext=phoneoutgoinglimit=1incominglimit=1canreinvite=nonat=yesqualify=yes [203]defaultuser=203secret=secret3type=peercallerid="user3 "host=dynamiccontext=phoneoutgoinglimit=1incominglimit=1canreinvite=nonat=yesqualify=yes [DID]context=sipdefaultuser=sipusernametype=peersecret=sippasswordhost=iptel.orgfromdomain=iptel.orgfromuser=253xxxxxxxtrustrpid = yessendrpid = yescanreinvite = noinsecure=port,invitenat=yes

extensions.conf

[general]static=nowriteprotect=noautofallthrough=yesclearglobalvars=yespriorityjumping=no [globals]gtimeout=50  ; timeout value; initializegvuser=10000 [sip]exten => _253xxxxxxx,1,ExecIf($[${gvuser}!=10000]?Bridge(${gvuser}):Dial(SIP/201&SIP/203,60,D(:1)))exten => _253xxxxxxx,n, Set(GLOBAL(gvuser)=10000)exten => _253xxxxxxx, n, Hangup()[phone]include => sipinclude => gv-outbound[gv-outbound]exten => _NXXNXXXXXX,1,GoTo(1${EXTEN},1)exten => _1NXXNXXXXXX,1,Answerexten => _1NXXNXXXXXX,n,Set(GLOBAL(gvuser)=${CHANNEL})exten => _1NXXNXXXXXX,n,System(gvoice -e gvusername@gmail.com -p gvpassword call ${EXTEN} 1253xxxxxxx 1 &)exten => _1NXXNXXXXXX,n,Ringingexten => _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, type help for list of cmds) some are: sip show peers sip show registry sip show users sip reload

Links

Phone setup How to use the 7060g DIrectory info

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