#egrep -v "^\;|^\ *$" /etc/asterisk/sip.conf
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
; Default is enabled
; defaults to "asterisk". If you set a system name in
; asterisk.conf, it defaults to that system name
; Realms MUST be globally unique according to RFC 3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
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
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
; international character conversions in URIs
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
; and subscriptions (seconds)
; Defaults to 100 ms
; fully. Enable this option to not get error messages
; when sending MWI to phones with this bug.
; Message-Account in the MWI notify message
; defaults to "asterisk"
; This may also be set for individual users/peers
; use 'never' to never use in-band signalling, even in cases
; where some buggy devices might not render it
; Valid values: yes, no, never Default: never
; Note that promiscredir when redirects are made to the
; local system will cause loops since Asterisk is incapable
; of performing a "hairpin" call.
; a valid phone number
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec -alaw, ulaw)
; auto : Use rfc2833 if offered, inband otherwise
; in the this section to get any video support at all.
; You can turn it off on a per peer basis if the general
; video support is enabled, but you can't enable it for
; one peer only without enabling in the general section.
; Videosupport and maxcallbitrate is settable
; for peers and users as well
; performs events (e.g. hold)
; for any reason, always reject with '401 Unauthorized'
; instead of letting the requester know whether there was
; a matching user or peer for their request
; order instead of RFC3551 packing order (this is required
; for Sipura and Grandstream ATAs, among others). This is
; contrary to the RFC3551 specification, the peer _should_
; be negotiating AAL2-G726-32 instead :-(
; your localnet setting. Unless you have some sort of strange network
; setup you will not need to enable this.
; on the audio channel
; when we're not on hold. This is to be able to hangup
; a call in the case of a phone disappearing from the net,
; like a powerloss or grandma tripping over a cable.
; on the audio channel
; when we're on hold (must be > rtptimeout)
; (default is off - zero)
; the moment the channel loads this configuration
; (see sip history / sip no history)
; SIP history is output to the DEBUG logging channel
; Useful to limit subscriptions to local extensions
; Settable per peer/user also
; Turning on notifyringing and notifyhold will add a lot
; more database transactions if you are using realtime.
; status notification when you are using type=friend
; Inbound calls, that really apply to the user part
; of a friend will now be added to and compared with
; the peer limit instead of applying two call limits,
; one for the peer and one for the user.
; "sip show inuse" will only show active calls on
; the peer side of a "type=friend" object if this
; setting is turned on.
; 0 = continue forever, hammering the other server
; until it accepts the registration
; Default is 0 tries, continue forever
; messages if we're behind a NAT
; The externip and localnet is used
; when registering and communicating with other proxies
; that we're registered with
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
; used
; You may add multiple local networks. A reasonable
; set of defaults are:
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581 (;rport)
; never = Never attempt NAT mode or RFC3581 support
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)
; RTP media stream (audio) to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of them is behind a NAT).
; The default setting is YES. If you have all clients
; behind a NAT, or for some other reason wants Asterisk to
; stay in the audio path, you may want to turn this off.
; In Asterisk 1.4 this setting also affect direct RTP
; at call setup (a new feature in 1.4 - setting up the
; call directly between the endpoints instead of sending
; a re-INVITE).
; the call directly with media peer-2-peer without re-invites.
; Will not work for video and cases where the callee sends
; RTP payloads and fmtp headers in the 200 OK that does not match the
; callers I