SIP电话Proxy守护神

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 protected] 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 protected] 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---