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doc/asterisk_sample.txt

@@ -1,236 +0,0 @@
-Example Setup, Asterisk running on the same machine as siproxd (= NAT host).
-
-I use a iptables rule to redirect all outgoing SIP traffix from Asterisk
-to siproxd.
-
-/etc/sysconfig/iptables:
-------------------------
-*nat
-######################################################################
-# NAT: redirect locally generated packets
-:OUTPUT - [0:0]
-#########################
-#
-# Asterisk Traffix via local siproxd. (must use DNAT to inbound IF! not REDIRECT)
--A OUTPUT -o ppp+ -p udp --sport 5061 -j DNAT  --to-destination 192.168.1.1:5060
-
-COMMIT
-
-
-
-
-/etc/asterisk/sip.conf:
------------------------
-[general]
-context = default
-allowoverlap = no  ; Disable overlap dialing support. (Default is yes)
-bindport = 5061    ; use a different port than 5060, as that port will be
-                   ; occupied by siproxd!
-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
-
-; g726 sounds very bad, useless!
-; g722 is a dead end, no conversion from/to possible
-; g729 sound like through a long metal tube
-disallow = all
-allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
-autoframing = yes
-
-allowexternaldomains = yes
-allowexternalinvites = yes
-allowguest = yes
-allowsubscribe = no
-allowtransfer = yes
-alwaysauthreject = no
-autodomain = yes
-callevents = no
-compactheaders = no
-dumphistory = no
-g726nonstandard = no
-ignoreregexpire = no
-jbenable = no
-jbforce = no
-jblog = no
-maxcallbitrate = 384
-maxexpiry = 3600
-minexpiry = 180
-notifyringing = no
-pedantic = no
-promiscredir = no
-recordhistory = no
-relaxdtmf = no
-rtcachefriends = no
-rtsavesysname = no
-rtupdate = no
-sendrpid = yes
-sipdebug = no
-t1min = 100
-progressinband = no
-t38pt_udptl = no
-trustrpid = no
-usereqphone = no
-videosupport = no
-;
-; the following is required when using siproxd with local DNAT rule
-nat=never
-externip=192.168.1.1
-;
-localnet = 192.168.0.0/16  ; my inbound network with local UAs
-domain = 192.168.1.1       ; inbound IP of host running Asterisk and siproxd
-domain = mynatfirewall     ; -"-
-canreinvite = no
-
-useragent = PBX            ; sipcall.ch (and others?) require UA string
-                           ; to be different from "AsteriskPBX"
-
-[authentication]
-;---end---
-
-
-/etc/asterisk/users.conf
-------------------------
-[general]
-;
-; Full name of a user
-;
-fullname = New User
-userbase = 200
-;
-; Create voicemail mailbox and use use macro-stdexten
-;
-hasvoicemail = yes
-;
-; Set voicemail mailbox 6000 password to 1234
-;
-vmsecret = 1234
-;
-; Create SIP Peer
-;
-hassip = yes
-hasiax = no
-;
-;
-; Create manager entry
-;
-hasmanager = no
-;
-; Remaining options are not specific to users.conf entries but are general.
-;
-callwaiting = yes
-threewaycalling = yes
-callwaitingcallerid = yes
-transfer = yes
-canpark = yes
-cancallforward = yes
-callreturn = yes
-callgroup = 1
-pickupgroup = 1
-host = dynamic
-localextenlength = 3
-allow_aliasextns = no
-allow_an_extns = no
-hasagent = no
-hasdirectory = no
-
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-; Local SIP UAs
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-[201]
-callwaiting = yes
-cid_number = 201
-context = local_sip
-email = email@host.xx
-fullname = Full Name
-group = 
-hasagent = yes
-hasdirectory = yes
-hasiax = no
-hasmanager = no
-hassip = yes
-hasvoicemail = yes
-host = dynamic
-mailbox = 201
-secret = <secret for AUTH>
-threewaycalling = yes
-zapchan = 
-registeriax = no
-registersip = yes
-vmsecret = <secret for voicemail>
-
-[202]
-callwaiting = yes
-cid_number = 202
-context = local_sip
-email = email@host.xx
-fullname = Full Name
-group = 
-hasagent = yes
-hasdirectory = yes
-hasiax = no
-hasmanager = no
-hassip = yes
-hasvoicemail = yes
-host = dynamic
-mailbox = 202
-secret = <secret for AUTH>
-threewaycalling = yes
-zapchan = 
-registeriax = no
-registersip = yes
-vmsecret = <secret for voicemail>
-
-
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-; SIP Trunks
-;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
-; sipphone.com
-[trunk_1]
-disallow = all
-allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
-callerid = 
-contact = 1747669xxxx
-context = DID_trunk_1
-dialformat = ${EXTEN:1}
-fromdomain = proxy01.sipphone.com
-fromuser = 1747669xxxx
-group = 
-hasexten = no
-hasiax = no
-hassip = yes
-host = proxy01.sipphone.com
-insecure = very
-port = 5060
-provider = 
-registeriax = no
-registersip = yes
-secret = <secret for AUTH>
-trunkname = Custom - sipphone1341
-trunkstyle = customvoip
-username = 1747669xxxx
-
-; sipcall.ch
-[trunk_3]
-disallow = all
-allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
-callerid = 
-contact = 4132511xxxx
-context = DID_trunk_3
-dialformat = ${EXTEN:1}
-fromdomain = sip.backbone.ch
-fromuser = 4132511xxxx
-group = 
-hasexten = no
-hasiax = no
-hassip = yes
-host = sip.backbone.ch
-insecure = very
-port = 5060
-provider = 
-registeriax = no
-registersip = yes
-secret = <secret for AUTH>
-trunkname = Custom - sipcall
-trunkstyle = customvoip
-username = 4132511xxxx
-
-;---end---