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Thomas Ries
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Example Setup, Asterisk running on the same machine as siproxd (= NAT host).
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I use a iptables rule to redirect all outgoing SIP traffix from Asterisk
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to siproxd.
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/etc/sysconfig/iptables:
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------------------------
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*nat
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######################################################################
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# NAT: redirect locally generated packets
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:OUTPUT - [0:0]
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#########################
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#
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# Asterisk Traffix via local siproxd. (must use DNAT to inbound IF! not REDIRECT)
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-A OUTPUT -o ppp+ -p udp --sport 5061 -j DNAT --to-destination 192.168.1.1:5060
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COMMIT
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/etc/asterisk/sip.conf:
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-----------------------
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[general]
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context = default
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allowoverlap = no ; Disable overlap dialing support. (Default is yes)
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bindport = 5061 ; use a different port than 5060, as that port will be
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; occupied by siproxd!
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bindaddr = 0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
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srvlookup = yes ; Enable DNS SRV lookups on outbound calls
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; g726 sounds very bad, useless!
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; g722 is a dead end, no conversion from/to possible
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; g729 sound like through a long metal tube
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disallow = all
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allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
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autoframing = yes
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allowexternaldomains = yes
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allowexternalinvites = yes
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allowguest = yes
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allowsubscribe = no
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allowtransfer = yes
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alwaysauthreject = no
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autodomain = yes
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callevents = no
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compactheaders = no
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dumphistory = no
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g726nonstandard = no
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ignoreregexpire = no
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jbenable = no
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jbforce = no
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jblog = no
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maxcallbitrate = 384
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maxexpiry = 3600
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minexpiry = 180
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notifyringing = no
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pedantic = no
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promiscredir = no
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recordhistory = no
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relaxdtmf = no
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rtcachefriends = no
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rtsavesysname = no
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rtupdate = no
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sendrpid = yes
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sipdebug = no
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t1min = 100
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progressinband = no
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t38pt_udptl = no
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trustrpid = no
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usereqphone = no
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videosupport = no
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;
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; the following is required when using siproxd with local DNAT rule
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nat=never
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externip=192.168.1.1
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;
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localnet = 192.168.0.0/16 ; my inbound network with local UAs
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domain = 192.168.1.1 ; inbound IP of host running Asterisk and siproxd
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domain = mynatfirewall ; -"-
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canreinvite = no
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useragent = PBX ; sipcall.ch (and others?) require UA string
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; to be different from "AsteriskPBX"
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[authentication]
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;---end---
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/etc/asterisk/users.conf
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------------------------
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[general]
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;
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; Full name of a user
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;
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fullname = New User
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userbase = 200
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;
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; Create voicemail mailbox and use use macro-stdexten
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;
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hasvoicemail = yes
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;
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; Set voicemail mailbox 6000 password to 1234
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;
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vmsecret = 1234
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;
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; Create SIP Peer
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;
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hassip = yes
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hasiax = no
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;
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;
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; Create manager entry
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;
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hasmanager = no
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;
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; Remaining options are not specific to users.conf entries but are general.
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;
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callwaiting = yes
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threewaycalling = yes
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callwaitingcallerid = yes
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transfer = yes
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canpark = yes
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cancallforward = yes
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callreturn = yes
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callgroup = 1
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pickupgroup = 1
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host = dynamic
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localextenlength = 3
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allow_aliasextns = no
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allow_an_extns = no
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hasagent = no
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hasdirectory = no
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;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
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; Local SIP UAs
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;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
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[201]
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callwaiting = yes
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cid_number = 201
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context = local_sip
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email = email@host.xx
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fullname = Full Name
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group =
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hasagent = yes
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hasdirectory = yes
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hasiax = no
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hasmanager = no
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hassip = yes
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hasvoicemail = yes
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host = dynamic
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mailbox = 201
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secret = <secret for AUTH>
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threewaycalling = yes
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zapchan =
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registeriax = no
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registersip = yes
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vmsecret = <secret for voicemail>
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[202]
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callwaiting = yes
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cid_number = 202
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context = local_sip
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email = email@host.xx
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fullname = Full Name
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group =
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hasagent = yes
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hasdirectory = yes
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hasiax = no
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hasmanager = no
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hassip = yes
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hasvoicemail = yes
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host = dynamic
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mailbox = 202
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secret = <secret for AUTH>
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threewaycalling = yes
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zapchan =
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registeriax = no
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registersip = yes
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vmsecret = <secret for voicemail>
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;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
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; SIP Trunks
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;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
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; sipphone.com
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[trunk_1]
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disallow = all
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allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
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callerid =
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contact = 1747669xxxx
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context = DID_trunk_1
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dialformat = ${EXTEN:1}
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fromdomain = proxy01.sipphone.com
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fromuser = 1747669xxxx
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group =
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hasexten = no
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hasiax = no
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hassip = yes
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host = proxy01.sipphone.com
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insecure = very
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port = 5060
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provider =
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registeriax = no
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registersip = yes
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secret = <secret for AUTH>
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trunkname = Custom - sipphone1341
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trunkstyle = customvoip
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username = 1747669xxxx
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; sipcall.ch
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[trunk_3]
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disallow = all
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allow = gsm,ulaw,alaw,adpcm,speex,g729,g723
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callerid =
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contact = 4132511xxxx
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context = DID_trunk_3
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dialformat = ${EXTEN:1}
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fromdomain = sip.backbone.ch
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fromuser = 4132511xxxx
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group =
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hasexten = no
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hasiax = no
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hassip = yes
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host = sip.backbone.ch
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insecure = very
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port = 5060
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provider =
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registeriax = no
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registersip = yes
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secret = <secret for AUTH>
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trunkname = Custom - sipcall
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trunkstyle = customvoip
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username = 4132511xxxx
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;---end---
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