VOIP trunks in Asterisk (Gizmo5, GTalk, VoiceStick and Stanaphone)

Add VOIP trunks by AsteriskNow GUI or edit users.conf manually. Type asterisk -r -vvv for more verbose debug information.

1. Gizmo5/Google Voice:
   [1sipnumber]
   context = DID_1sipnumber
   host = proxy01.sipphone.com
   trunkname = Gizmo5 ; GUI metadata
   username = 1sipnumber
   secret = password
   hasiax = no
   registeriax = no
   hassip = yes
   registersip = yes
   trunkstyle = voip
   hasexten = no
   canreinvite = yes
   disallow = all
   qualify = yes
   allow = ulaw,alaw
   insecure = port,invite
   
   The context can be found in the file extensions.conf as below
   [DID_1sipnumber]
   exten = s,1,GotoIf($[${LEN(${CALLERID(num)})} > 10]?1-setcid,1)
   exten = s,n,Goto(1-dial,1)
   exten = 1-setcid,1,Set(CALLERID(num)=${CALLERID(num):2})
   exten = 1-setcid,n,Goto(1-dial,1)
   exten = 1-dial,1,Goto(default,6000,1)
   exten = 1-dial,n,Hangup()
   
   The above context strips the leading "+1" from the incoming caller ID the provider(sipphone) sends to Asterisk and rings extension 6000 for the incoming calls.
   
   Ref: How to change incoming CallerID
   


2. VoiceStick (avoid it if possible): it uses outbound proxy 72.5.80.116:5060 or 72.5.80.117:80. But I couldn't make it work with my Asterisk or Linksys PAP2T under their Next2Nothing or Asterisk Two plan.
   Add the following to /etc/hosts
   72.5.80.116 i2telecom.com
   and the trunk in users.conf
   [1phonenumber]
   context=DID_1phonenumber
   host=i2telecom.com
   trunkname=i2telecom.com
   username=1phonenumber
   secret=password
   hasiax=no
   registeriax=no
   hassip=yes
   registersip=yes
   trunkstyle=voip
   hasexten=no
   disallow=all
   allow=all
   qualify = yes
   canreinvite = no
   insecure = port,invite
   


3. GTalk:
   I have the following in the file extensions.conf to set the correct incoming caller ID for the google account that's calling in. The name of the caller will be shown as Gtalk/google_account_name
   [gtalk-in]
   exten = _.,1,NoOp(${CHANNEL})
   exten = _.,2,Set(CALLERID(name)=${CUT(CHANNEL,,1)})
   exten = _.,3,Set(CALLERID(num)=${CUT(CHANNEL,,2)})
   exten = _.,4,Goto(default,6000,1)
   exten = _.,5,Hangup()
   
   
4. Stanaphone: It's important to have the right insecure setting. Otherwise it will try Digest-MD5 authentication for incoming calls and fail instantly.
   [username]
   context = DID_username
   host = sip.stanaphone.com
   trunkname = Stanaphone ; GUI metadata
   username = username
   secret = password
   hasiax = no
   registeriax = no
   hassip = yes
   registersip = yes
   trunkstyle = voip
   hasexten = no
   disallow = all
   allow = all
   qualify = yes
   canreinvite = yes
   insecure = port,invite