sip.conf settings for Asterisk 1.8- [general]
- ; Global Settings
- bindport = 5060 ; Port to bind to (SIP is 5060)
- bindaddr = 0.0.0.0 ; Bind all addresses on machine
- realm = asterisk
- useragent = myuseragent
- ;sdpsession = myuseragent
- allowguest = yes ; Allow or reject guest calls ; set allowguest = no for security reason
- allowsubscribe = yes
- canreinvite = no
- insecure = port,invite
- srvlookup = yes
- ;qualifyfreq = 60
- ;qualifygap = 100
- ;qualifypeers = 1
- callevents = no
- ;allowexternalinvites = yes
- allowexternaldomains = yes
- alwaysauthreject = yes
- allowoverlap = no ; Disable overlap dialing support (Default is yes)
- allowtransfer = yes ; Disable all transfers
- videosupport = no
- callcounter = yes
- t38pt_udptl = yes,fec,maxdatagram = 400
- faxdetect = yes
- ; Network QoS Settings
- tos_sip = CS3 ; Sets TOS for SIP packets
- tos_audio = ef ; Sets TOS for RTP audio packets.
- tos_video = AF41 ; Sets TOS for RTP video packets
- cos_sip = 3
- cos_audio = 5
- cos_video = 4
- cos_text = 3
- jbenable = no
- jbforce = no
- ; Network Settings
- externrefresh = 10
- externhost = your_ddns_name ; DDNS
- fromdomain = your_ddns_name ; Optional - force a particular domain
- localnet = xxx.xxx.xxx.xxx/255.255.255.0 ; Asterisk network address and mask
- stunaddr =
- autodomain = no
- ; Global Signaling Settings
- disallow = all
- allow = ulaw
- allow = alaw
- allow = gsm ; GSM needs low bandwidth than ulaw and alaw
- allow = g729
- allow = slin
- faxdetect = on
- rtptimeout = 60
- rtpholdtimeout = 300
- rtpkeepalive = 20 ; Send a keepalive ever 20 Seconds if using NAT
- maxexpiry = 3600 ; **Engin & BBP Global this if necessary
- minexpiry = 60
- defaultexpiry = 240 ; **Engin users: include users: include this if necessary
- registerattempts = 0
- registertimeout = 20
- relaxdtmf = yes
- notifyringing = yes
- notifyhold = yes
- notifycid = yes
- pedantic = no
- progressinband = never
- promiscredir = no
- ; Default Settings
- nat = yes
- dtmfmode = rfc2833
- qualify = yes
- context = default ; Send unknown SIP incoming callers to this context
- language = en
- musicclass = default
- mohinterpret = default
- mohsuggest = default
- [authentication]
- [my-settings](!) ; template for the phones
- type = friend ; is both peer (out) and user (in)
- qualify = yes
- nat = yes
- host = dynamic
- dtmfmode = auto
- allow = ulaw,alaw,gsm,g729,slin
- context = yourcontextname
- canreinvite = no ; set "canreinvite = yes" for all internal extensions. This will let RTP directly flow from Line 1 or other ATA to PSTN without going through Asterisk, thus to minimize the delay
- insecure = port,invite
- port = 5060
- ;musiconhold = default
- ;musciclass = default
- ;deny=0.0.0.0/0.0.0.0
- ;permit=xxx.xxx.xxx/255.255.255.0
- [6001](my-settings)
- defaultuser = 6001
- secret = very_secret_code
- mailbox = 6001@default
- vmsecret = 6001
- dial = SIP/6001
- callerid = "who_is_who" <>
- ;accountcode =
- ;callgroup = 1,3-4 ; members of groups 1,3 to 4
- ;pickupgroup = 1,2-4 ; member of "pickup" groups 1,2 to 4
- call-limit = 10
- musiconhold = default
- musciclass = default
- [6002](phone-settings)
- defaultuser = 6002
- secret = very_secret_code
- mailbox = 6002@default
- vmsecret = 6002
- dial = SIP/6002
- callerid = "who_am_i" <>
- ;accountcode =
- ;callgroup = 1,3-4 ; members of groups 1,3 to 4
- ;pickupgroup = 1,2-4 ; member of "pickup" groups 1,2 to 4
- call-limit = 10
- musiconhold = friends
- musciclass = friends
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發表於 2011-9-29 22:54
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