ASL-Corp members, Bureaucrats, Check users, Structured Discussions bots, oversight, Suppressors, Administrators, Upload Wizard campaign editors
1,714
edits
Changes
Jump to navigation
Jump to search
Created initial page
The following configuration will allow you to dial into your AllStarLink/PTTLink node and use it like your own personal telephone portal.
==Prerequisites==
*A SIP provider to setup a trunk to
*A provisioned telephone number (DID) with your provider that is passed over the trunk to your node
*Forwarding of port 5060 from your cable modem/router to your AllStarLink/PTTLink node
It is recommended that you setup firewall rules to limit inbound connections on port 5060 to only your SIP provider. This will eliminate the SIP enumeration and other attacks that your ALlStarLink/PTTLink node will be subjected to.
===SIP Providers===
The following SIP providers have been used with this configuration.
Note: You will only need a single provider.
*[https://www.leap.tel/ Leap]<ref name="LEAP">[https://www.leap.tel Leap]/</ref>
*[https://telnyx.com/ Telnyx]<ref name="TELNYX">[https://telnyx.com/ Telnyx]</ref>
==Configuration==
Make the following changes to your AllStarLink/PTTLink node's configuration files.
===sip.conf===
Edit /etc/asterisk/sip.conf as follows:
====Using TELNYX====
Add a stanza to sip.conf similar to the following:
[TELNYX]
type=friend
host=sip.telnyx.com
disallow=all
allow=ulaw
insecure=private
context=from-ptsn
canreinvite=yes
qualify=no
username='''USERNAME'''
secret='''SECRET'''
nat=yes
directtypdrtup=yes
externalip='''YOUR PUBLIC IP''' ;useful if you have issues with SIP
;localhost=NETWORK/SUBNET ;useful if you have issues with SIP
====Using Leap====
Add a stanza to sip.conf similar to the following:
[LEAP]
disallow=all
allow=ulaw
context=from-pstn
type=friend
insecure=invite
dtmfmode=rfc2833
username='''USERNAME'''
secret='''PASSWORD'''
host='''FQDN OF YOUR LEAP TEL INSTANCE NAME'''
fromdomain='''MAKE THIS MATCH THE HOST LINE ABOVE'''
canreinvite=yes
nat=yes
qualify=yes
externip='''YOUR PUBLIC IP'''
localnet='''YOUR NODE'S LOCAL NEWORK/SUBNET''' ; Format: 192.168.0.0/255.255.255.0
===SIP registration===
You will need to add a register line in the [general] section of your sip.conf file. This will allow your node to authenticate with your SIP provider and optionally use it for outbound calls.
====Telnyx====
register => <username>:<password>@sip.telnyx.com
====Leap====
register => <username>:<password>@<fqdn of leap instance>
===extensions.conf===
We use the context of '''from-pstn''' in both SIP providers examples above. You will need to add a dial plan stanza that matches the context calls come in on your SIP trunk.
'''''DO NOT PUSH EXTERNAL PHONE CALLS TO YOU NODE'S DEFAULT STANZA!'''''
This is a security concern as all calls coming in should be authenticated before they are passed to app_rpt.
You can change the greeting message and tones played to whatever you want.
We will use the node number of ''2000'' as the extension to dial to get node access. You will be greeted by a message to enter your password and if successful the call will then transfer over to your node.
[from-pstn]
exten => XXXXXXXXXX,1,Goto(s|1) ; Replace XXXXXXXXXX with the full phone number passed from your provider (e.g., 2535551212)
exten => s,1,Ringing()
exten => s,n,Wait(3)
exten => s,n,Answer()
exten => s,n,Log(VERBOSE,Incoming call from ${CALLERID(all)}) ;Log the incoming call
exten => s,n,GotoIf($["${CALLERID(num)}" = ""]?877) ;Do some spam/telemarketer filtering
exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "877"]?877)
exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "800"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?877)
exten => s,n,Wait(2)
exten => s,n,Background(thank-you-for-calling&privacy-please-stay-on-line-to-be-connected)
exten => s,n,Background(or&if-u-know-ext-dial&otherwise&to-hang-up-2)
exten => s,n,Background(press-star)
exten => s,n,Wait(1)
exten => s,n,Background()
exten => s,n,Playtones(425/50,0/50)
exten => s,n,Playtones(!914/276,!1371/276,!1777/380,0)
exten => s,n,Wait(2)
exten => s,n,WaitExten(20) ; Wait for a keypress
exten => s,n,Playback(goodbye) ;Nothing received so hangup
exten => s,n,Wait(2)
exten => s,n,Hangup()
;Default of where calls will go if nothing is entered
exten => 10,1,LOG(VERBOSE, Hanging up.)
exten => 10,n,StopPlaytones
exten => 10,n,Playback(goodbye)
exten => 10,n,Wait(2)
exten => 10,n,Hangup()
exten => *,1,Playback(goodbye)
exten => *,n,Wait(2)
exten => *,n,Hangup()
;Drop calls with no caller ID, 800, or 877
exten => s,877,Congestion()
exten => 877,1,Wait(1)
exten => 877,n,Hangup()
exten => i,1,Log(VERBOSE, Caller ${CALLERID(all)} dialed an invalid extension. Hanging up...)
exten => i,n,Congestion(15)
exten => h,1,Log(VERBOSE, Caller ${CALLERID(all)} hung up.)
exten => h,n,Hangup()
exten => 2000,1,Log(VERBOSE, Caller ${CALLERID(all)} is attempting remote radio control....)
exten => 2000,n,Authenticate('''1234''') ; Change 1234 to a more secure password
exten => 2000,n,Goto(pstn-radio-control|2000|1)
Now we add an additional stanza to the extensions.conf called pstn-radio-control. This stanza will take the phone call we passed to it and connect it to app_rpt.
[pstn-radio-control]
exten => 2000,1,Ringing
exten => 2000,n,Wait(3)
exten => 2000,n,Answer
exten => 2000,n,Set(CALLERID(name)="'''CALLSIGN'''") ; Change to your callsign
exten => 2000,n,Playback(rpt/connected)
exten => 2000,n,Playback(rpt/node)
exten => 2000,n,Saydigits(${EXTEN})
exten => 2000,n,Rpt(${EXTEN}|P|${CALLERID(name)})
==Testing==
*Make sure your AllStarLink/PTTLink node is registered with your SIP provider.
Node*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.telnyx.com:5060 blahblah 105 Registered Sat, 17 Jul 2021 12:35:43
*Call the telephone number you have provisioned with your SIP provider and watch your AllStarLink/PTTLink node's console. If successful you will see output similar to:
-- Executing [18005551212@from-pstn:1] Goto("SIP/LEAP-01aa5b48", "s|1") in new stack
-- Goto (from-pstn,s,1)
-- Executing [s@from-pstn:1] Ringing("SIP/LEAP-01aa5b48", "") in new stack
-- Executing [s@from-pstn:2] Wait("SIP/LEAP-01aa5b48", "3") in new stack
-- Executing [s@from-pstn:3] Answer("SIP/LEAP-01aa5b48", "") in new stack
-- Executing [s@from-pstn:4] Log("SIP/LEAP-01aa5b48", "VERBOSE|Incoming call from "WIRELESS CALLER" <+18885551212>") in new stack
*You will hear the greeting, be prompted to enter the extension you wish to call and hear some tones playing.
*Enter your node number (you did change it from 2000 in the example right?) and your password when prompted.
*If successful you will then be passed to app_rpt and hear the announcement saying you've connected to your node, your call sign as the other node connecting, and your node will now be ready to control using DTMF codes as if you were on a radio link.
==References/External Links==
<references />
==Prerequisites==
*A SIP provider to setup a trunk to
*A provisioned telephone number (DID) with your provider that is passed over the trunk to your node
*Forwarding of port 5060 from your cable modem/router to your AllStarLink/PTTLink node
It is recommended that you setup firewall rules to limit inbound connections on port 5060 to only your SIP provider. This will eliminate the SIP enumeration and other attacks that your ALlStarLink/PTTLink node will be subjected to.
===SIP Providers===
The following SIP providers have been used with this configuration.
Note: You will only need a single provider.
*[https://www.leap.tel/ Leap]<ref name="LEAP">[https://www.leap.tel Leap]/</ref>
*[https://telnyx.com/ Telnyx]<ref name="TELNYX">[https://telnyx.com/ Telnyx]</ref>
==Configuration==
Make the following changes to your AllStarLink/PTTLink node's configuration files.
===sip.conf===
Edit /etc/asterisk/sip.conf as follows:
====Using TELNYX====
Add a stanza to sip.conf similar to the following:
[TELNYX]
type=friend
host=sip.telnyx.com
disallow=all
allow=ulaw
insecure=private
context=from-ptsn
canreinvite=yes
qualify=no
username='''USERNAME'''
secret='''SECRET'''
nat=yes
directtypdrtup=yes
externalip='''YOUR PUBLIC IP''' ;useful if you have issues with SIP
;localhost=NETWORK/SUBNET ;useful if you have issues with SIP
====Using Leap====
Add a stanza to sip.conf similar to the following:
[LEAP]
disallow=all
allow=ulaw
context=from-pstn
type=friend
insecure=invite
dtmfmode=rfc2833
username='''USERNAME'''
secret='''PASSWORD'''
host='''FQDN OF YOUR LEAP TEL INSTANCE NAME'''
fromdomain='''MAKE THIS MATCH THE HOST LINE ABOVE'''
canreinvite=yes
nat=yes
qualify=yes
externip='''YOUR PUBLIC IP'''
localnet='''YOUR NODE'S LOCAL NEWORK/SUBNET''' ; Format: 192.168.0.0/255.255.255.0
===SIP registration===
You will need to add a register line in the [general] section of your sip.conf file. This will allow your node to authenticate with your SIP provider and optionally use it for outbound calls.
====Telnyx====
register => <username>:<password>@sip.telnyx.com
====Leap====
register => <username>:<password>@<fqdn of leap instance>
===extensions.conf===
We use the context of '''from-pstn''' in both SIP providers examples above. You will need to add a dial plan stanza that matches the context calls come in on your SIP trunk.
'''''DO NOT PUSH EXTERNAL PHONE CALLS TO YOU NODE'S DEFAULT STANZA!'''''
This is a security concern as all calls coming in should be authenticated before they are passed to app_rpt.
You can change the greeting message and tones played to whatever you want.
We will use the node number of ''2000'' as the extension to dial to get node access. You will be greeted by a message to enter your password and if successful the call will then transfer over to your node.
[from-pstn]
exten => XXXXXXXXXX,1,Goto(s|1) ; Replace XXXXXXXXXX with the full phone number passed from your provider (e.g., 2535551212)
exten => s,1,Ringing()
exten => s,n,Wait(3)
exten => s,n,Answer()
exten => s,n,Log(VERBOSE,Incoming call from ${CALLERID(all)}) ;Log the incoming call
exten => s,n,GotoIf($["${CALLERID(num)}" = ""]?877) ;Do some spam/telemarketer filtering
exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "877"]?877)
exten => s,n,GotoIf($["${CALLERID(num):0:3}" = "800"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:9}" = "Anonymous"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:7}" = "Unknown"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:7}" = "Private"]?877)
exten => s,n,GotoIf($["${CALLERIDNAME:0:7}" = "Private"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:10}" = "Restricted"]?877)
exten => s,n,GotoIf($["${CALLERIDNUM:0:4}" = "PSTN"]?877)
exten => s,n,Wait(2)
exten => s,n,Background(thank-you-for-calling&privacy-please-stay-on-line-to-be-connected)
exten => s,n,Background(or&if-u-know-ext-dial&otherwise&to-hang-up-2)
exten => s,n,Background(press-star)
exten => s,n,Wait(1)
exten => s,n,Background()
exten => s,n,Playtones(425/50,0/50)
exten => s,n,Playtones(!914/276,!1371/276,!1777/380,0)
exten => s,n,Wait(2)
exten => s,n,WaitExten(20) ; Wait for a keypress
exten => s,n,Playback(goodbye) ;Nothing received so hangup
exten => s,n,Wait(2)
exten => s,n,Hangup()
;Default of where calls will go if nothing is entered
exten => 10,1,LOG(VERBOSE, Hanging up.)
exten => 10,n,StopPlaytones
exten => 10,n,Playback(goodbye)
exten => 10,n,Wait(2)
exten => 10,n,Hangup()
exten => *,1,Playback(goodbye)
exten => *,n,Wait(2)
exten => *,n,Hangup()
;Drop calls with no caller ID, 800, or 877
exten => s,877,Congestion()
exten => 877,1,Wait(1)
exten => 877,n,Hangup()
exten => i,1,Log(VERBOSE, Caller ${CALLERID(all)} dialed an invalid extension. Hanging up...)
exten => i,n,Congestion(15)
exten => h,1,Log(VERBOSE, Caller ${CALLERID(all)} hung up.)
exten => h,n,Hangup()
exten => 2000,1,Log(VERBOSE, Caller ${CALLERID(all)} is attempting remote radio control....)
exten => 2000,n,Authenticate('''1234''') ; Change 1234 to a more secure password
exten => 2000,n,Goto(pstn-radio-control|2000|1)
Now we add an additional stanza to the extensions.conf called pstn-radio-control. This stanza will take the phone call we passed to it and connect it to app_rpt.
[pstn-radio-control]
exten => 2000,1,Ringing
exten => 2000,n,Wait(3)
exten => 2000,n,Answer
exten => 2000,n,Set(CALLERID(name)="'''CALLSIGN'''") ; Change to your callsign
exten => 2000,n,Playback(rpt/connected)
exten => 2000,n,Playback(rpt/node)
exten => 2000,n,Saydigits(${EXTEN})
exten => 2000,n,Rpt(${EXTEN}|P|${CALLERID(name)})
==Testing==
*Make sure your AllStarLink/PTTLink node is registered with your SIP provider.
Node*CLI> sip show registry
Host Username Refresh State Reg.Time
sip.telnyx.com:5060 blahblah 105 Registered Sat, 17 Jul 2021 12:35:43
*Call the telephone number you have provisioned with your SIP provider and watch your AllStarLink/PTTLink node's console. If successful you will see output similar to:
-- Executing [18005551212@from-pstn:1] Goto("SIP/LEAP-01aa5b48", "s|1") in new stack
-- Goto (from-pstn,s,1)
-- Executing [s@from-pstn:1] Ringing("SIP/LEAP-01aa5b48", "") in new stack
-- Executing [s@from-pstn:2] Wait("SIP/LEAP-01aa5b48", "3") in new stack
-- Executing [s@from-pstn:3] Answer("SIP/LEAP-01aa5b48", "") in new stack
-- Executing [s@from-pstn:4] Log("SIP/LEAP-01aa5b48", "VERBOSE|Incoming call from "WIRELESS CALLER" <+18885551212>") in new stack
*You will hear the greeting, be prompted to enter the extension you wish to call and hear some tones playing.
*Enter your node number (you did change it from 2000 in the example right?) and your password when prompted.
*If successful you will then be passed to app_rpt and hear the announcement saying you've connected to your node, your call sign as the other node connecting, and your node will now be ready to control using DTMF codes as if you were on a radio link.
==References/External Links==
<references />