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- ;
- ; SIP Configuration for Asterisk
- ;
- ; Syntax for specifying a SIP device in extensions.conf is
- ; SIP/devicename where devicename is defined in a section below.
- ;
- ; You may also use
- ; SIP/username@domain to call any SIP user on the Internet
- ; (Don't forget to enable DNS SRV records if you want to use this)
- ;
- ; If you define a SIP proxy as a peer below, you may call
- ; SIP/proxyhostname/user or SIP/user@proxyhostname
- ; where the proxyhostname is defined in a section below
- ;
- ; Useful CLI commands to check peers/users:
- ; sip show peers Show all SIP peers (including friends)
- ; sip show users Show all SIP users (including friends)
- ; sip show registry Show status of hosts we register with
- ;
- ; sip debug Show all SIP messages
- ;
- ; module reload chan_sip.so Reload configuration file
- ; Active SIP peers will not be reconfigured
- ;
- ;------- Naming devices ------------------------------------------------------
- ;
- ; When naming devices, make sure you understand how Asterisk matches calls
- ; that come in.
- ; 1. Asterisk checks the SIP From: address username and matches against
- ; names of devices with type=user
- ; The name is the text between square brackets [name]
- ; 2. Asterisk checks the IP address (and port number) that the INVITE
- ; was sent from and matches against any devices with type=peer
- ;
- ; Don't mix extensions with the names of the devices. Devices need a unique
- ; name. The device name is *not* used as phone numbers. Phone numbers are
- ; anything you declare as an extension in the dialplan (extensions.conf).
- ;
- ; Note: The parameter "username" is not the username and in most cases is
- ; not needed at all. Check below. In later releases, it's renamed
- ; to "defaultuser" which is a better name, since it is used in
- ; combination with the "defaultip" setting.
- ;-----------------------------------------------------------------------------
- [general]
- context=unauthenticated
- allowguest=no
- allowoverlap=no
- bindport=5060
- bindaddr=0.0.0.0
- srvlookup=yes
- tos_sip=cs3 ; Sets TOS for SIP packets.
- tos_audio=ef ; Sets TOS for RTP audio packets.
- disallow=all
- allow=ulaw
- allow=g729
- dtmfmode=rfc2833
- alwaysauthreject=yes
- register => 105245:2FsuonGrOuq4r@chicago.voip.ms
- externip=200.201.202.203
- ;externhost=some.hostname.com
- localnet=192.168.192.0/255.255.255.0
- localnet=169.254.0.0/255.255.0.0
- nat=yes
- directmedia=nonat
- directmediadeny=0.0.0.0/0 ; Use directmediapermit and directmediadeny to restrict
- directmediapermit=192.168.192.0/24 ; which peers should be able to pass directmedia to each other
- [ipkall] ; ITSP - static, no register statement
- type=peer
- disallow=all
- allow=ulaw
- ;allow=g729 ; Uncomment if you support G729
- context=ipkall-inbound
- dtmfmode=rfc2833
- host=66.54.140.46
- [ipkall1](ipkall) ; ITSP - static, no register statement
- host=66.54.140.47
- [voipms] ; ITSP - dynamic, register statement used above
- type=peer
- defaultuser=105245
- remotesecret=2FsuonGrOuq4r
- trustrpid=yes
- sendrpid=yes
- host=chicago.voip.ms
- ;port=5080 ; alternate port
- ;port=42872 ; alternate port
- dtmfmode=rfc2833
- deny=0.0.0.0/0.0.0.0
- permit=64.120.22.242/255.255.255.255
- context=voipms-inbound
- directmedia=no
- disallow=all
- allow=ulaw
- [00001234FFFF-a] ; This phone will use extension 123
- type=peer
- context=phones
- callerid=Testing Phone <123>
- secret=testing123
- host=dynamic
- accountcode=123
- directmedia=no
- qualify=yes
- disallow=all
- allow=ulaw
- mailbox=123@default
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