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Netw320 Lab 5-6

In: Computers and Technology

Submitted By bigjohnlapp457
Words 1252
Pages 6
NETW320
Week 5 and week 6 – Asterisk* Lab

Here You can see that the address is set and that the Ports are now active.

And I can get a line for 3 seconds Using my windows phone but it doesn’t stay connected. I have now been working on getting tested with Chas from class

This is the coding that I used to set up my Config line. I am still working on the configuration setup.This was taken out of the Red Hat Book.
John's extensions.conf

[general] static=no writeprotect=no autofallthrough=yes clearglobalvars=yes priorityjumping=no

[globals] ; Incoming calls to these numbers will be routed to the specified peer. peer-5555551234 = SIP/mango peer-5555554321 = SIP/mangosteen ; Incoming calls to these numbers will be routed to the specified context. context-8775551234 = DISA-in,s,1 ; These names will override whatever is sent as the Caller ID Name for incoming calls. callerid-5555556789 = Joe Blow ; These are the peers outgoing calls may be routed through, in order. If one is unreachable, Asterisk will attempt to use the next one. outgoing-1 = VoIPmsUS3 outgoing-2 = Callcentric ; This is the password for use with DISA. disa-pass = 1234567

[devices] ; When a device wishes to place a call, Asterisk will attempt to match the dialed number in the following contexts, in this order: include => special include => outgoing

; We use this context for anything that should not simply dial out via a VoIP provider. [special] ; This is the call recorder. To use, conference extension 200 into your call. exten => 200,1,Answer() ; Note that it may or may not be legal to record a telephone call in your area. Check with your local authorities if you are unsure. exten => 200,n,Playback(beep) ; Record any calls to this extension in u-law format. You may record in wav as well but it requires more system resources. exten => 200,n,MixMonitor(/opt/recordings/${UNIQUEID}.ulaw) ; Record for a maximum of 15 minutes. exten => 200,n,Wait(900) exten => 200,n,Playback(beep) exten => 200,n,Hangup() ; Music-on-Hold test. exten => 202,1,Answer() exten => 202,n,MusicOnHold() exten => 202,n,Hangup() ; For Canadians: if 1-800 (1-866, 1-877, 1-888) is dialed, route through a Canadian carrier. This plan does not match 800 (without the ; leading 1) so they will be routed through the default (in our case American) carrier. exten => _18[0678][0678]NXXXXXX,1,Dial(SIP/${EXTEN}@VoIPmsCA2) ; This will also match other non-TF area codes such as 867 but that's fine. exten => _18[0678][0678]NXXXXXX,n,Hangup

; This is the context which allows devices to place outgoing calls. [outgoing] exten => _[*#0-9].,1,NoOp ; Try to route the call internally first. Does context-${EXTEN} exist? exten => _[*#0-9].,n,GotoIf($[${LEN(${context-${EXTEN}})}!=0]?${context-${EXTEN}}); ; Does ${peer-[${EXTEN}} exist? exten => _[*#0-9].,n,GotoIf($[${LEN(${peer-${EXTEN}})}==0]?external) exten => _[*#0-9].,n,Dial(${peer-${EXTEN}}) ; We were not able to route the call internally. So, route the call with a VoIP provider. Start with the first peer in the list. exten => _[*#0-9].,n(external),Set(peer=1) ; trypeer: If the status is UNREACHABLE or UNAVAILABLE, go to failover. (You must set qualify=yes in the peer's context in sip.conf for ; this to work.) exten => _[*#0-9].,n(trypeer),GotoIf($["${SIPPEER(${outgoing-${peer}},status)}"="UNREACHABLE"|"${SIPPEER(${outgoing-${peer}},status)}"="UNAVAILABLE"]?failover) ; Attempt to send the call out through ${outgoing-${peer}}. exten => _[*#0-9].,n,Dial(SIP/${EXTEN}@${outgoing-${peer}}) ; If it didn't work, go to failover. exten => _[*#0-9].,n,NoOp(${DIALSTATUS}) exten => _[*#0-9].,n,GotoIf($["${DIALSTATUS}"="CHANUNAVAIL"|"${DIALSTATUS}"="CONGESTION"]?failover) exten => _[*#0-9].,n,Hangup ; failover: Increment peer by one. If outgoing-${peer} exists, go to trypeer. exten => _[*#0-9].,n(failover),Set(peer=$[${peer}+1]) exten => _[*#0-9].,n,GotoIf($[${LEN(${outgoing-${peer}})}!=0]?trypeer) ; If we're here, then we were not able to place the call through any peer. Play a message to the caller saying that outgoing lines ; are unavailable. exten => _[*#0-9].,n,Playback(all-outgoing-lines-unavailable) exten => _[*#0-9].,n,Hangup ; Some possible ${DIALSTATUS} values: ; Improperly formatted number -> CHANUNAVAIL ; Unable to reach server -> CHANUNAVAIL ; bad password in sip.conf -> CONGESTION ; Some possible ${SIPPEER(${outgoing-${peer}},status)} values: ; Improperly formatted number -> OK (36ms) ; Bad password in sip.conf -> OK (36ms) ; Unable to reach server -> UNREACHABLE ; qualify=no -> Unmonitored

; This is the context where we route incoming calls from VoIP providers. [incoming] exten => _NXXNXXXXXX,1,Ringing ; Override the Caller ID, if necessary. exten => _NXXNXXXXXX,n,Set(CALLERID(name)=${IF($[${LEN(${callerid-${CALLERID(number)}})}!=0]?${callerid-${CALLERID(number)}}:${CALLERID(name)})}) ; Capitalize the first letter of each word in the Caller ID. This is purely for cosmetic purposes. Requires gnu-sed; use ipkg install sed ; if you're running Asterisk on Tomato. exten => _NXXNXXXXXX,n,Set(CALLERID(name)=${SHELL(echo -n ${CALLERID(name)} | gnu-sed 's/\(\w\)\([^ ]*\)/\U\1\L\2/g')}) ; Go to another context, if necessary. This is used for DISA. You could also use it for an IVR. exten => _NXXNXXXXXX,n,GotoIf($[${LEN(${context-${EXTEN}})}!=0]?${context-${EXTEN}}); ; The only thing left to try is a peer. exten => _NXXNXXXXXX,n,GotoIf($[${LEN(${peer-${EXTEN}})}=0]?notfound) exten => _NXXNXXXXXX,n,Dial(${peer-${EXTEN}}) ; Neither a peer nor a context is configured for ${EXTEN}. Claim congestion. exten => _NXXNXXXXXX,n(notfound),Congestion

; Incoming calls may be routed to this context. If they are, Asterisk will ask them to enter their password. [DISA-in] exten => s,1,Set(TIMEOUT(response)=20) exten => s,n,Set(TIMEOUT(digit)=20) ; For some reason, after you enter the password, Asterisk waits for the digit timeout, not the response timeout, for you to begin entering the number you wish to dial. If the number you have dialed cannot be matched by any other extension in the DISA-out context, Asterisk will dial immediately without waiting the 20 seconds. exten => s,n,Wait(.5) exten => s,n,Playback(after-the-tone) exten => s,n,Playback(enter-password) ; Give the user a dial tone and let them dial a number as defined by DISA-out. exten => s,n,DISA(${disa-password},DISA-out) ; This is the context that DISA users are allowed to access. [DISA-out] ; Go to the context-${EXTEN} context, if it exists. exten => _NXXNXXXXXX,1,GotoIf($[${LEN(${context-${EXTEN}})}!=0]?${context-${EXTEN}}); ; If there is no variable named peer-${EXTEN}, then go to the disa-external context. exten => _NXXNXXXXXX,n,GotoIf($["${LEN(${peer-${EXTEN}})}"="0"]?disa-external) ; I couldn't get internal calls to ring, so I play this message: exten => _NXXNXXXXXX,n,Playback(pls-wait-connect-call) exten => _NXXNXXXXXX,n,Dial(${peer-${EXTEN}}, 18) ; If the call wasn't answered, go to the disa-external context so we'll be routed to the externally-hosted voicemail. exten => _NXXNXXXXXX,n,GotoIf($["${DIALSTATUS}"!="ANSWER"]?disa-external) exten => _NXXNXXXXXX,n,Hangup ; Use Transfer() instead of Dial() so that we do not proxy the audio. ${CHANNEL(peerip)} is the IP of the VoIP server. exten => _NXXNXXXXXX,n(disa-external),Transfer(SIP/${EXTEN}@${CHANNEL(peerip)}) exten => _1NXXNXXXXXX,1,Transfer(SIP/${EXTEN}@${CHANNEL(peerip)})

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