[asterisk-users] INVITE Sending Local IP

Mike Bessette admin at 6jonline.com
Thu Oct 1 05:15:54 CDT 2009

OK. Here is the relevant section of my sip.conf

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=		; IP address to bind to ( 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
				; Add domain and configure incoming context
				; for external calls to this domain
;domain=			; 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.
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
;       register => user[:secret[:authuser]]@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:password at mysipprovider.com
<1234%3Apassword at mysipprovider.com>
;     This will pass incoming calls to the 's' extension
;register => 2345:password at sip_proxy/1234

register => 193*****36:passw0rd******@did.voip.les.net/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 =	; 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=; All RFC 1918 addresses are local networks
;localnet=	; Also RFC1918
;localnet=		; Another RFC1918 with CIDR notation
;localnet= ;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.

; 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:topsecret at digium.com <mark%3Atopsecret at digium.com>
; 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 <steve at geekinter.net> 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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