[arm-allstar] SIP
Chris Andrist
chris.andrist at outlook.com
Wed Mar 3 22:39:47 EST 2021
Hi Chris,
I would load the asterisk cli and run the following command:
sip debug level 7
Then dial into the phone number and see what happens on the command line.
—
Regards,
Chris Andrist, KC7WSU
> On Mar 3, 2021, at 7:14 PM, Chris Viningre via ARM-allstar <arm-allstar at hamvoip.org> wrote:
>
> I'm trying to configure hamvoip to receive incoming calls. I'm using voip.ms
> and my extension for my node is registered on voip.ms. When I call my DID
> number and enter the extension it hangs up. If I call the extension from my
> phone it goes to a busy signal.
>
> extension.conf with passwords x out
> [general]
>
> static=yes ; These two lines prevent the command-line interface
> writeprotect=yes ; from overwriting the config file. Leave them here.
>
> [globals]
> HOMENPA=915
>
> [default]
> exten => i,1,Hangup
>
> [radio-secure]
> exten => 43934,1,rpt,43934
> exten => 46039,1,rpt,46039
>
> [radio-secure-proxy]
> exten => 43934,1,rpt,43934|X
> exten => _0X.,1,Goto(allstar-sys|${EXTEN:1}|1)
>
> ; IAXRPT Windows
>
> [radio-iaxrpt]
> exten=43934,1,Rpt,43934|X
> exten=46039,1,Rpt,46039|X
>
> ; The following stanza is to be used for Android/Iphone
> ; connections. Please configure the node number for the
> ; node on this server you want to connect to.
> ; The callerID name is configured and returned from
> ; your phone. It is typically your call.
>
> [phone-iaxrpt] ;;; Stanza is the context from iax.conf
> exten => 43934,1,Answer
> exten => 43934,n,Playback,rpt/node
> exten => 43934,n,Playback,digits/4
> exten => 43934,n,Playback,digits/3
> exten => 43934,n,Playback,digits/9
> exten => 43934,n,Playback,digits/3
> exten => 43934,n,Playback,digits/4
> exten => 43934,n,Set(CALLERID(num)=0)
> exten => 43934,n,Rpt,43934|P|${CALLERID(name)} ;;; The "CallerID" from
> IAXRpt
>
> [radio-connect]
>>>> exten => 595,1,Wait(1)
>>>> exten => 595,n,Dial(SIP/telephone_dialin:Mbikes3#@192.168.1.169/43934)
>>>> exten => 595,n,Hangup
>>>>
>>>> exten => 596,1,Wait(1)
>>>> exten => 596,n,Dial(SIP/
>>>> telephone_dialin:224273_43934:xxxxx at 192.168.1.169:4568/43934
>>>> exten => 596,n,Hangup
>>>>
>>>> And this stanza is in my Allstar iax.conf on BOTH of the servers at
> the
>>>> local IP addresses above.
>>>>
>>>> [telephone_dialin]
>>>> type=user
>>>> secret=xxxxx
>>>> disallow=all
>>>> allow=ulaw
>>>> allow=g726aal2
>>>> allow=gsm
>>>> codecpriority=host
>>>> context=radio-control
>>>> transfer=no
>
> [radio-control]
>>>> exten=43934,1,Answer
>>>> exten=43934,n,Playback(rpt/node)
>>>> exten=43934,n,Playback(digits/4)
>>>> exten=43934,n,Playback(digits/3)
>>>> exten=43934,n,Playback(digits/9)
>>>> exten=43934,n,Playback(digits/3)
>>>> exten=43934,n,Playback(digits/4)
>>>> exten=43934,n,Playback(rpt/connected)
>>>> exten=43934,n,Rpt,43934|Pv
>
> ; To make this have a static callerID name if your phone
> ; app does not support callerID change the last line to this
> ; and specify the name (usually CALL) in place of static-name.
>
> ;exten => 43934,n,Rpt,1998|P|"static-name"
>
> ; Autopatch example lines. Must be configured and
> ; SIP or IAX phone connections configured to work.
>
> [pstn-out]
> exten=_NXXNXXXXXX,1,playback(ss-noservice)
> exten=_NXXNXXXXXX,2,Congestion
>
> [invalidnum]
> exten=s,1,Wait,3
> exten=s,n,Playback,ss-noservice
> exten=s,n,Wait,1
> exten=s,n,Hangup
>
> [radio]
> exten=_X11,1,Goto(check_route|${EXTEN}|1);
> exten=_NXXXXXX,1,Goto(check_route|1${HOMENPA}${EXTEN}|1)
> exten=_1XXXXXXXXXX,1,Goto(check_route|${EXTEN}|1)
>
> [check_route]
> ; no 800
> exten=_1800NXXXXXX,2,Goto(invalidnum|s|1)
> exten=_1888NXXXXXX,2,Goto(invalidnum|s|1)
> exten=_1877NXXXXXX,2,Goto(invalidnum|s|1)
> exten=_1866NXXXXXX,2,Goto(invalidnum|s|1)
> exten=_1855NXXXXXX,2,Goto(invalidnum|s|1)
> ; no X00 NPA
> exten=_1X00XXXXXXX,2,Goto(invalidnum|s|1)
> ; no X11 NPA
> exten=_1X11XXXXXXX,2,Goto(invalidnum|s|1)
> ; no X11
> exten=_X11,2,Goto(invalidnum|s|1)
> ; no 555 Prefix in any NPA
> exten=_1NXX555XXXX,2,Goto(invalidnum|s|1)
> ; no 976 Prefix in any NPA
> exten=_1NXX976XXXX,2,Goto(invalidnum|s|1)
> ; no NPA=809
> exten=_1809XXXXXXX,2,Goto(invalidnum|s|1)
> ; no NPA=900
> exten=_1900XXXXXXX,2,Goto(invalidnum|s|1)
>
> ; okay, route it
> exten=_1NXXXXXXXXX,1,Goto(pstn-out|${EXTEN:1}|1)
> exten=_X.,2,Goto(invalidnum|s|1)
>
> ; End autopatch example
>
> ; Following stanza required for web transceiver access
>
> [allstar-public]
>
> exten => s,1,Ringing
> exten => s,n,Set(RESP=${CURL(
> https://register.allstarlink.org/cgi-bin/authwebphone.pl?${CALLERID(name)})
> })
> exten => s,n,Set(NODENUM=${CALLERID(number)})
> exten => s,n,GotoIf($["${RESP:0:1}" = "?"]?hangit)
> exten => s,n,GotoIf($["${RESP:0:1}" = ""]?hangit)
> exten => s,n,GotoIf($["${RESP:0:5}" != "OHYES"]?hangit)
> exten => s,n,Set(CALLSIGN=${RESP:5})
> exten => s,n,Wait(3)
> exten => s,n,Playback(rpt/node|noanswer)
> exten => s,n,Saydigits(${NODENUM})
> exten => s,n,Set(CALLERID(num)=0)
> exten => s,n,Set(CALLERID(name)=${CALLSIGN})
> exten => s,n,Rpt(${NODENUM}|X)
> exten => s,n,Hangup
> exten => s,n(hangit),Answer
> exten => s,n(hangit),Wait(1)
> exten => s,n(hangit),Hangup
>
> exten => 9999,1,Goto(allstar-public|s|1)
>
> ; end web transceiver stanza
>
> ; Following stanza required for phone access
>
> [allstar-sys]
>
> exten => _x.,1,Ringing
> exten => _x.,n,Wait(3)
> exten => _x.,n,Answer
> exten => _x.,n,Playback(rpt/node)
> exten => _x.,n,Saydigits(${EXTEN:1})
> exten => _x.,n,Rpt(${EXTEN:1}|P|${CALLERID(name)}-P)
> exten => _x.,n,Hangup
>
> ; end phone access stanza
>
> ; The blacklist and whitelist stanzas below are used to allow or ban nodes
> ; from connecting. Only one can be selected at a time as defined in the
> ; [radio] stanza of iax.conf
> ;
> ; The best way to allow or ban nodes from the database is to
> ; use the Supermon application or the node-ban-allow.sh script.
> ;
> ; No changes need to be made to these stanzas. See comments in iax.conf.
>
> [blacklist]
> exten => _XXXX!,1,NoOp(${CALLERID(num)})
> exten =>
> _XXXX!,n,GotoIf($[${DB_EXISTS(blacklist/${CALLERID(num)})}]?blocked)
> exten => _XXXX!,n,Goto(radio-secure,${EXTEN},1)
> exten => _XXXX!,n(blocked),Hangup
>
> [whitelist]
> exten => _XXXX!,1,NoOp(${CALLERID(num)})
> exten => _XXXX!,n,NoOp(${IAXPEER(CURRENTCHANNEL)})
> exten => _XXXX!,n,GotoIf($["${IAXPEER(CURRENTCHANNEL)}" =
> "127.0.0.1"]?radio-secure,${EXTEN},1) ;permit local IPs
> exten =>
> _XXXX!,n,GotoIf($[${DB_EXISTS(whitelist/${CALLERID(num)})}]?radio-secure,${EXTEN},1)
> exten => _XXXX!,n,Hangup
>
> ; Stanza to get node number and pass to saydns.sh script in rpt.conf
> [saydns]
> exten => _xxx.,1,System(/usr/local/sbin/saydns.sh ${EXTEN})
> exten => _xxx.,n,Hangup()
>
> ; Example connection to another Asterisk server
> ; and passing an extension
>
> ; [pbx_server]
> ;exten => _1NXXNXXXXXX,1,Dial(IAX2/pbx/${EXTEN})
> ;exten => _NXXNXXXXXX,1,Dial(IAX2/pbx/${EXTEN})
> ;exten => _NXX,1,Dial(IAX2/pbx/${EXTEN})
> ;exten => _NX,1,Dial(IAX2/pbx/${EXTEN})
>
> #includeifexists custom/extensions.conf
>
> [mycontext]
> ; Make sure to include inbound prior to outbound because the _NXXNXXXXXX
> handler will match the incoming call and create a loop
> include => voipms-inbound
> include => voipms-outbound
>
> [voipms-outbound]
> exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
> exten => _1NXXNXXXXXX,n,Hangup()
> exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
> exten => _NXXNXXXXXX,n,Hangup()
> exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
> exten => _011.,n,Hangup()
> exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
> exten => _00.,n,Hangup()
>
> ; inbound context example for your DID numbers, do not add the number 1 in
> front
>
> [voipms-inbound]
> exten => 9152014604,1,Answer() ;your DID
>
> sip.conf with password x out
>
> [general]
> register => 224273_43934:xxxxx at dallas2.voip.ms:5060
>
> context=default ; 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)
> bindaddr=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
> ; Note: Asterisk only uses the first host
> ; in SRV records
> ; Disabling DNS SRV lookups disables the
> ; ability to place SIP calls based on domain
> ; names to some other SIP users on the Internet
>
> [voipms]
> canreinvite=nonat
> context=mycontext
> host=dallas2.voip.ms ;(one of our multiple servers, you can choose the one
> closer to your location)
> type=peer
> disallow=all
> allow=ulaw
> ; allow=g729 ; uncomment if you support g729
> nat=yes
>
>
>
> --
> Chris Viningre
> Website: ws5b.com <http://ws5b.com>
> Email: chris at ws5b.com <chris at ws5b.com>
> _______________________________________________
>
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