Merge branch 'feature/improved-configuration'
This commit is contained in:
commit
425a29a27a
38
asterisk/extensions.conf
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38
asterisk/extensions.conf
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[+441234567890]
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; Redirect incoming calls from trunk with context +441234567890 to internal extension 0.
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exten => _+X.,1,Goto(internal,0,1)
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[outgoing]
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; Redirect all calls entering this context via SIP trunk "twilio".
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exten => _X.,1,Dial(SIP/twilio/+${EXTEN})
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same => n,Hangup
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[internal]
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; Dialplan for extension 1000.
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; Dial SIP user 1000. Timeout after 10 seconds.
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exten => 1000,1,Dial(SIP/1000,10)
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same => n,VoiceMail(${EXTEN})
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same => n,Hangup
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; Dialplan for extension 1001.
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exten => 1001,1,Dial(SIP/1001,10)
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same => n,VoiceMail(${EXTEN})
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same => n,Hangup
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; Dialgroup "everyone" at extension 0 to ring mulitple extensions.
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exten => 0,1,Set(DIALGROUP(everyone,add)=SIP/1000)
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same => n,Set(DIALGROUP(everyone,add)=SIP/1001)
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same => n,Dial(${DIALGROUP(everyone)},10)
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same => n,VoiceMail(${EXTEN})
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same => n,Hangup
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; Dialplans for redirecting calls to outgoing context.
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; Redirect international calls in format 00441234567890.
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exten => _00X.,1,Goto(outgoing,${EXTEN:2},1)
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; Redirect international calls in format +441234567890.
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exten => _+X.,1,Goto(outgoing,${EXTEN:1},1)
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; Redirect outgoing national calls in format 01234567890.
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exten => _0ZX.,1,Goto(outgoing,44${EXTEN:1},1)
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2
asterisk/modules.conf
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2
asterisk/modules.conf
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[modules]
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autoload=yes
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35
asterisk/sip.conf
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35
asterisk/sip.conf
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[general]
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; Default context for incoming calls.
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context=public
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; Determines whether a phone is available before accepting call.
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qualify=yes
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; Settings common to all trunks.
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[trunk](!)
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type=peer
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nat=no
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qualify=yes
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insecure=port,invite
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dtmfmode=rfc2833
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allow=ulaw,alaw,G729
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; Settings common to all internal extensions.
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[internal](!)
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context=internal
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type=friend
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host=dynamic
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allow=ulaw,alaw,G729
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; Twilio trunk.
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[twilio](trunk)
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context=+441234567890
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fromuser=+441234567890
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host=example.pstn.ie1.twilio.com
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; Internal extensions.
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[1000](internal)
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secret=supersecurepassword
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[1001](internal)
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secret=supersecurepassword
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12
asterisk/voicemail.conf
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12
asterisk/voicemail.conf
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[general]
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format=wav
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maxsilence=10
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maxmessage=180
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[default]
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; Voicemail inbox for missed dialgroup calls.
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0 => 0000,0,test@example.com
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; Voicemail inboxes for individual extensions.
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1000 => 0000,1000,test@example.com
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1001 => 0000,1001,test@example.com
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@ -1,12 +0,0 @@
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[general]
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autofallthrough=yes
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[globals]
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[nexmo-trunk]
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exten => _X.,1,Dial(SIP/${EXTEN},30)
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[internal]
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exten => 666,1,Dial(SIP/666)
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exten => 777,1,Dial(SIP/777)
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exten => _X.,1,Dial(SIP/${EXTEN}@nexmo-<phone_number>)
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@ -1,2 +0,0 @@
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[modules]
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autoload=yes
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51
nexmo.conf
51
nexmo.conf
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[general]
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context=nexmo-trunk
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register => <username>:<api_key>@sip.nexmo.com
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nat=yes
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[nexmo]
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context=nexmo-trunk
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fromdomain=sip.nexmo.com
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type=peer
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insecure=port,invite
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nat=no
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dtmfmode=rfc2833
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allow=ulaw
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allow=alaw
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allow=G729
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[nexmo-sip-01](nexmo)
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host=5.10.112.121
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[nexmo-sip-02](nexmo)
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host=5.10.112.122
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[nexmo-sip-03](nexmo)
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host=119.81.44.6
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[nexmo-sip-04](nexmo)
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host=119.81.44.7
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[nexmo-sip-05](nexmo)
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host=169.60.141.29
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[nexmo-sip-06](nexmo)
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host=169.60.141.30
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[nexmo-<phone_number>]
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context=nexmo-trunk
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username=<username>
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defaultuser=<username>
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secret=<api_key>
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fromuser=<phone_number>
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fromdomain=sip.nexmo.com
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host=sip.nexmo.com
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type=peer
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insecure=very
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qualify=yes
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nat=no
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dtmfmode=rfc2833
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allow=ulaw
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allow=alaw
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allow=G729
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