OK. Here is the relevant section of my sip.conf
[general]
context=default ; Default context for incoming calls
;allowguest=no ; Allow or reject guest calls (default is yes,
this
can also be set to 'osp'
; if asterisk was compiled with OSP support.
realm=windsorwebdynamic.com ; Realm for digest authentication
; defaults to "asterisk"
; 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)
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
domain=windsorwebdynamic.com ; Set default domain for this host
; If configured, Asterisk will only allow
; INVITE and REFER to non-local domains
; Use "sip show domains" to list local domains
domain=windsorwebdynamic.com
; Add domain and configure incoming context
; for external calls to this domain
;domain=1.2.3.4 ; Add IP address as local domain
; You can have several "domain" settings
allowexternalinvites=yes ; Disable INVITE and REFER to non-local domains
; Default is yes
;autodomain=yes ; Turn this on to have Asterisk add local host
; name and local IP to domain list.
;pedantic=yes ; Enable slow, pedantic checking for Pingtel
; and multiline formatted headers for strict
; SIP compatibility (defaults to "no")
;tos=184 ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay ; lowdelay,throughput,reliability,mincost,none
;maxexpiry=3600 ; Max length of incoming registration we allow
;defaultexpiry=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
; Message-Account in the MWI
notify message
; defaults to "asterisk"
;videosupport=yes ; Turn on support for SIP video
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ;
;musicclass=default ; Sets the default music on hold class for all
SIP calls
; This may also be set for individual
users/peers
;language=en ; Default language setting for all users/peers
; This may also be set for individual
users/peers
;relaxdtmf=yes ; Relax dtmf handling
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP
activity
; when we're not on hold
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP
activity
; when we're on hold (must be > rtptimeout)
;trustrpid = no ; If Remote-Party-ID should be trusted
;sendrpid = yes ; If Remote-Party-ID should be sent
;progressinband=never ; If we should generate in-band ringing always
; 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
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP
address
; Note that promiscredir when redirects are
made to the
; local system will cause loops since SIP is
incapable
; of performing a "hairpin" call.
;usereqphone = no ; If yes, ";user=phone" is added to uri that
contains
; a valid phone number
;dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF.
Default: rfc2833
; Other options:
; info : SIP INFO messages
; inband : Inband audio (requires 64 kbit codec
-alaw, ulaw)
; auto : Use rfc2833 if offered, inband
otherwise
;compactheaders = yes ; send compact sip headers.
;sipdebug = yes ; Turn on SIP debugging by default, from
; the moment the channel loads this
configuration
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
; Useful to limit subscriptions to local
extensions
; Settable per peer/user also
;notifyringing = yes ; Notify subscriptions on RINGING state
;
; If regcontext is specified, Asterisk will dynamically create and destroy a
; NoOp priority 1 extension for a given peer who registers or unregisters with
; us. The actual extension is the 'regexten' parameter of the registering
; peer or its name if 'regexten' is not provided. More than one regexten may
; be supplied if they are separated by '&'. Patterns may be used in regexten.
;
;regcontext=sipregistrations
;
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuse...@host[:port][/extension]
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:[email protected]
<1234%[email protected]>
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:passw...@sip_proxy/1234
register => 193*****36:passw0rd***[email protected]/193*****36
;
; Register 2345 at sip provider 'sip_proxy'. Calls from this provider
; connect to local extension 1234 in extensions.conf, default context,
; unless you configure a [sip_proxy] section below, and configure a
; context.
; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com]
; Tip 2: Use separate type=peer and type=user sections for SIP providers
; (instead of type=friend) if you have calls in both directions
;registertimeout=20 ; retry registration calls every 20 seconds
(default)
;registerattempts=10 ; Number of registration attempts before we
give up
; 0 = continue forever, hammering the other
server until it
; accepts the registration
; Default is 0 tries, continue forever
;callevents=no ; generate manager events when sip ua performs
events
(e.g. hold)
;----------------------------------------- NAT SUPPORT ------------------------
; The externip, externhost and localnet settings are used if you use Asterisk
; behind a NAT device to communicate with services on the outside.
externip = 69.168.165.164 ; Address that we're going to put in
outbound SIP 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
;externhost=foo.dyndns.net ; Alternatively you can specify an
; external host, and Asterisk will
; perform DNS queries periodically. Not
; recommended for production
; environments! Use externip instead
;externrefresh=10 ; How often to refresh externhost if
; used
; You may add multiple local networks. A
reasonable set of defaults
; are:
;localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
;localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
;localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
; The nat= setting is used when Asterisk is on a public IP, communicating with
; devices hidden behind a NAT device (broadband router). If you have one-way
; audio problems, you usually have problems with your NAT configuration or your
; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP
; ports for incoming audio in rtp.conf
;
nat=yes ; Global NAT settings (Affects all peers and
users)
; yes = Always ignore info and assume NAT
; no = Use NAT mode only according to RFC3581
; never = Never attempt NAT mode or
RFC3581 support
; route = Assume NAT, don't send rport
; (work around more UNIDEN bugs)
;rtcachefriends=yes ; Cache realtime friends by adding them to the
internal list
; just like friends added from the config file
only on a
; as-needed basis? (yes|no)
;rtupdate=yes ; Send registry updates to database using
realtime? (yes|no)
; If set to yes, when a SIP UA registers
successfully, the ip address,
; the origination port, the registration
period, and the username of
; the UA will be set to database via realtime.
If not present,
defaults to 'yes'.
;rtautoclear=yes ; Auto-Expire friends created on the
fly on the same schedule
; as if it had just registered?
(yes|no|<seconds>)
; If set to yes, when the registration expires,
the friend will vanish from
; the configuration until requested again. If
set to an integer,
; friends expire within this number of seconds
instead of the
; registration interval.
;ignoreregexpire=yes ; Enabling this setting has two functions:
;
; For non-realtime peers, when their
registration expires, the information
; will _not_ be removed from memory or the
Asterisk database; if you attempt
; to place a call to the peer, the existing
information will be
used in spite
; of it having expired
;
; For realtime peers, when the peer is
retrieved from realtime storage,
; the registration information will be used
regardless of whether
; it has expired or not; if it expires while
the realtime peer is still in
; memory (due to caching or other reasons), the
information will not be
; removed from realtime storage
; Incoming INVITE and REFER messages can be matched against a list of 'allowed'
; domains, each of which can direct the call to a specific context if desired.
; By default, all domains are accepted and sent to the default context or the
; context associated with the user/peer placing the call.
; Domains can be specified using:
; domain=<domain>[,<context>]
; Examples:
; domain=myasterisk.dom
; domain=customer.com,customer-context
;
; In addition, all the 'default' domains associated with a server should be
; added if incoming request filtering is desired.
; autodomain=yes
;
; To disallow requests for domains not serviced by this server:
; allowexternaldomains=no
fromdomain=windsorwebdynamic.com ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.
[authentication]
; Global credentials for outbound calls, i.e. when a proxy challenges your
; Asterisk server for authentication. These credentials override
; any credentials in peer/register definition if realm is matched.
;
; This way, Asterisk can authenticate for outbound calls to other
; realms. We match realm on the proxy challenge and pick an set of
; credentials from this list
; Syntax:
; auth = <user>:<secret>@<realm>
; auth = <user>#<md5secret>@<realm>
; Example:
;auth=mark:[email protected] <mark%[email protected]>
;
; You may also add auth= statements to [peer] definitions
; Peer auth= override all other authentication settings if we match on realm
On Thu, Oct 1, 2009 at 5:56 AM, Steve Howes <[email protected]> wrote:
> On 1 Oct 2009, at 10:43, Mike Bessette wrote:
> > Hello. I set up an Asterisk box a couple days ago and was having
> > problems with not being able to hear SIP clients. After some
> > troubleshooting we have determined that hte INVITE is sending my
> > local(192.168) IP. How would I get * to send the public IP instead
> > of the local one? I have changed every IP/domain setting in sip.conf
> > to reflect my public IP but it still doesnt want to work. Thanks to
> > anyone hthat can help me.
>
> If you show us your config so we can see what is wrong...
>
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