[arm-allstar] SIP

[David McGough]

Hi Chris,

Do you have all the ports forwarded that are needed for SIP to work?

voip.ms also supports IAX2 and that's what I always use. Then you don't 
need chan_sip at all, unless you're also connecting local phones to the 
network.

73, David KB4FXC



On Wed, 3 Mar 2021, "Chris Viningre via ARM-allstar" wrote:

> Should the IP in the extensions.conf be the local IP or the Public IP?
> 
> On Wed, Mar 3, 2021 at 7:01 PM Chris Viningre <chris at ws5b.com> 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>
> >
> 
> 
>