OK. Here is the relevant section of my sip.conf<br><br><pre>[general]<br>context=default Default context for incoming calls<br>;allowguest=no Allow or reject guest calls (default is yes, this can also be set to 'osp'<br>
if asterisk was compiled with OSP support.<br>realm=<a href="http://windsorwebdynamic.com">windsorwebdynamic.com</a> Realm for digest authentication<br> defaults to "asterisk"<br> Realms MUST be globally unique according to RFC 3261<br>
Set this to your host name or domain name<br>bindport=5060 UDP Port to bind to (SIP standard port is 5060)<br>bindaddr=0.0.0.0 IP address to bind to (0.0.0.0 binds to all)<br>srvlookup=yes Enable DNS SRV lookups on outbound calls<br>
Note: Asterisk only uses the first host <br> in SRV records<br> Disabling DNS SRV lookups disables the <br> ability to place SIP calls based on domain <br> names to some other SIP users on the Internet<br>
<br>domain=<a href="http://windsorwebdynamic.com">windsorwebdynamic.com</a> Set default domain for this host<br> If configured, Asterisk will only allow<br> INVITE and REFER to non-local domains<br> Use "sip show domains" to list local domains<br>
domain=<a href="http://windsorwebdynamic.com">windsorwebdynamic.com</a><br> Add domain and configure incoming context<br> for external calls to this domain<br>;domain=1.2.3.4 Add IP address as local domain<br>
You can have several "domain" settings<br>allowexternalinvites=yes Disable INVITE and REFER to non-local domains<br> Default is yes<br>;autodomain=yes Turn this on to have Asterisk add local host<br>
name and local IP to domain list.<br>;pedantic=yes Enable slow, pedantic checking for Pingtel<br> and multiline formatted headers for strict<br> SIP compatibility (defaults to "no")<br>;tos=184 Set IP QoS to either a keyword or numeric val<br>
;tos=lowdelay lowdelay,throughput,reliability,mincost,none<br>;maxexpiry=3600 Max length of incoming registration we allow<br>;defaultexpiry=120 Default length of incoming/outoing registration<br>;notifymimetype=text/plain Allow overriding of mime type in MWI NOTIFY<br>
;checkmwi=10 Default time between mailbox checks for peers<br>;vmexten=voicemail ; dialplan extension to reach mailbox sets the <br> Message-Account in the MWI notify message <br> defaults to "asterisk"<br>
;videosupport=yes Turn on support for SIP video<br>;recordhistory=yes Record SIP history by default <br> (see sip history / sip no history)<br><br>;disallow=all First disallow all codecs<br>;allow=ulaw Allow codecs in order of preference<br>
;allow=ilbc <br>;musicclass=default Sets the default music on hold class for all SIP calls<br> This may also be set for individual users/peers<br>;language=en Default language setting for all users/peers<br>
This may also be set for individual users/peers<br>;relaxdtmf=yes Relax dtmf handling<br>;rtptimeout=60 Terminate call if 60 seconds of no RTP activity<br> when we're not on hold<br>;rtpholdtimeout=300 Terminate call if 300 seconds of no RTP activity<br>
when we're on hold (must be > rtptimeout)<br>;trustrpid = no If Remote-Party-ID should be trusted<br>;sendrpid = yes If Remote-Party-ID should be sent<br>;progressinband=never If we should generate in-band ringing always<br>
use 'never' to never use in-band signalling, even in cases<br> where some buggy devices might not render it<br> Valid values: yes, no, never Default: never<br>;useragent=Asterisk PBX Allows you to change the user agent string<br>
;promiscredir = no If yes, allows 302 or REDIR to non-local SIP address<br> Note that promiscredir when redirects are made to the<br> local system will cause loops since SIP is incapable<br>
of performing a "hairpin" call.<br>;usereqphone = no If yes, ";user=phone" is added to uri that contains<br> a valid phone number<br>;dtmfmode = rfc2833 Set default dtmfmode for sending DTMF. Default: rfc2833<br>
Other options: <br> info : SIP INFO messages<br> inband : Inband audio (requires 64 kbit codec -alaw, ulaw)<br> auto : Use rfc2833 if offered, inband otherwise<br><br>;compactheaders = yes send compact sip headers.<br>
;sipdebug = yes Turn on SIP debugging by default, from<br> the moment the channel loads this configuration<br>;subscribecontext = default Set a specific context for SUBSCRIBE requests<br> Useful to limit subscriptions to local extensions<br>
Settable per peer/user also<br>;notifyringing = yes Notify subscriptions on RINGING state<br><br>;<br>; If regcontext is specified, Asterisk will dynamically create and destroy a<br>; NoOp priority 1 extension for a given peer who registers or unregisters with<br>
; us. The actual extension is the 'regexten' parameter of the registering<br>; peer or its name if 'regexten' is not provided. More than one regexten may<br>; be supplied if they are separated by '&'. Patterns may be used in regexten.<br>
;<br>;regcontext=sipregistrations<br>;<br>; Asterisk can register as a SIP user agent to a SIP proxy (provider)<br>; Format for the register statement is:<br>; register => user[:secret[:authuser]]@host[:port][/extension]<br>
;<br>; If no extension is given, the 's' extension is used. The extension needs to<br>; be defined in extensions.conf to be able to accept calls from this SIP proxy<br>; (provider).<br>;<br>; host is either a host name defined in DNS or the name of a section defined<br>
; below.<br>;<br>; Examples:<br>;<br>;register => <a href="mailto:1234%3Apassword@mysipprovider.com">1234:password@mysipprovider.com</a> <br>;<br>; This will pass incoming calls to the 's' extension<br>;<br>
;<br>;register => 2345:password@sip_proxy/1234<br><br>register => 193*****36:passw0rd******@<a href="http://did.voip.les.net/193*****36">did.voip.les.net/193*****36</a><br><br>;<br>; Register 2345 at sip provider 'sip_proxy'. Calls from this provider<br>
; connect to local extension 1234 in extensions.conf, default context,<br>; unless you configure a [sip_proxy] section below, and configure a<br>; context.<br>; Tip 1: Avoid assigning hostname to a sip.conf section like [<a href="http://provider.com">provider.com</a>]<br>
; Tip 2: Use separate type=peer and type=user sections for SIP providers<br>; (instead of type=friend) if you have calls in both directions<br> <br>;registertimeout=20 retry registration calls every 20 seconds (default)<br>
;registerattempts=10 Number of registration attempts before we give up<br> 0 = continue forever, hammering the other server until it <br> accepts the registration<br> Default is 0 tries, continue forever<br>
;callevents=no generate manager events when sip ua performs events (e.g. hold)<br><br>;----------------------------------------- NAT SUPPORT ------------------------<br>; The externip, externhost and localnet settings are used if you use Asterisk<br>
; behind a NAT device to communicate with services on the outside.<br><br>externip = 69.168.165.164 Address that we're going to put in outbound SIP messages<br> if we're behind a NAT<br><br> The externip and localnet is used<br>
when registering and communicating with other proxies<br> that we're registered with<br>;externhost=<a href="http://foo.dyndns.net">foo.dyndns.net</a> Alternatively you can specify an <br> external host, and Asterisk will <br>
perform DNS queries periodically. Not<br> recommended for production <br> environments! Use externip instead<br>;externrefresh=10 How often to refresh externhost if <br> used<br> You may add multiple local networks. A reasonable set of defaults<br>
are:<br>;localnet=<a href="http://192.168.0.0/255.255.0.0">192.168.0.0/255.255.0.0</a>; All RFC 1918 addresses are local networks<br>;localnet=<a href="http://10.0.0.0/255.0.0.0">10.0.0.0/255.0.0.0</a> Also RFC1918<br>
;localnet=<a href="http://172.16.0.0/12">172.16.0.0/12</a> Another RFC1918 with CIDR notation<br>;localnet=<a href="http://169.254.0.0/255.255.0.0">169.254.0.0/255.255.0.0</a> ;Zero conf local network<br><br>; The nat= setting is used when Asterisk is on a public IP, communicating with<br>
; devices hidden behind a NAT device (broadband router). If you have one-way<br>; audio problems, you usually have problems with your NAT configuration or your<br>; firewall's support of SIP+RTP ports. You configure Asterisk choice of RTP<br>
; ports for incoming audio in rtp.conf<br>;<br>nat=yes Global NAT settings (Affects all peers and users)<br> ; yes = Always ignore info and assume NAT<br> ; no = Use NAT mode only according to RFC3581 <br>
; never = Never attempt NAT mode or RFC3581 support<br> route = Assume NAT, don't send rport <br> (work around more UNIDEN bugs)<br><br>;rtcachefriends=yes Cache realtime friends by adding them to the internal list<br>
just like friends added from the config file only on a<br> as-needed basis? (yes|no)<br><br>;rtupdate=yes Send registry updates to database using realtime? (yes|no)<br> If set to yes, when a SIP UA registers successfully, the ip address,<br>
the origination port, the registration period, and the username of<br> the UA will be set to database via realtime. If not present, defaults to 'yes'.<br><br>;rtautoclear=yes Auto-Expire friends created on the fly on the same schedule<br>
as if it had just registered? (yes|no|<seconds>)<br> If set to yes, when the registration expires, the friend will vanish from<br> the configuration until requested again. If set to an integer,<br> friends expire within this number of seconds instead of the<br>
registration interval.<br><br>;ignoreregexpire=yes Enabling this setting has two functions:<br> <br> For non-realtime peers, when their registration expires, the information<br> will _not_ be removed from memory or the Asterisk database; if you attempt<br>
to place a call to the peer, the existing information will be used in spite<br> of it having expired<br> <br> For realtime peers, when the peer is retrieved from realtime storage,<br> the registration information will be used regardless of whether<br>
it has expired or not; if it expires while the realtime peer is still in<br> memory (due to caching or other reasons), the information will not be<br> removed from realtime storage<br><br>; Incoming INVITE and REFER messages can be matched against a list of 'allowed'<br>
; domains, each of which can direct the call to a specific context if desired.<br>; By default, all domains are accepted and sent to the default context or the<br>; context associated with the user/peer placing the call.<br>
; Domains can be specified using:<br>; domain=<domain>[,<context>]<br>; Examples:<br>; domain=myasterisk.dom<br>; domain=<a href="http://customer.com">customer.com</a>,customer-context<br>;<br>; In addition, all the 'default' domains associated with a server should be<br>
; added if incoming request filtering is desired.<br>; autodomain=yes<br>;<br>; To disallow requests for domains not serviced by this server:<br>; allowexternaldomains=no<br><br> fromdomain=<a href="http://windsorwebdynamic.com">windsorwebdynamic.com</a> ; When making outbound SIP INVITEs to<br>
; non-peers, use your primary domain "identity"<br> ; for From: headers instead of just your IP<br> ; address. This is to be polite and<br>
; it may be a mandatory requirement for some<br> ; destinations which do not have a prior<br> ; account relationship with your server. <br><br>[authentication]<br>
; Global credentials for outbound calls, i.e. when a proxy challenges your<br>; Asterisk server for authentication. These credentials override<br>; any credentials in peer/register definition if realm is matched.<br>;<br>
; This way, Asterisk can authenticate for outbound calls to other<br>; realms. We match realm on the proxy challenge and pick an set of <br>; credentials from this list<br>; Syntax:<br>; auth = <user>:<secret>@<realm><br>
; auth = <user>#<md5secret>@<realm><br>; Example:<br>;auth=<a href="mailto:mark%3Atopsecret@digium.com">mark:topsecret@digium.com</a><br>; <br>; You may also add auth= statements to [peer] definitions <br>
; Peer auth= override all other authentication settings if we match on realm<br><br></pre><br><br><div class="gmail_quote">On Thu, Oct 1, 2009 at 5:56 AM, Steve Howes <span dir="ltr"><<a href="mailto:steve@geekinter.net">steve@geekinter.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="h5">On 1 Oct 2009, at 10:43, Mike Bessette wrote:<br>
> Hello. I set up an Asterisk box a couple days ago and was having<br>
> problems with not being able to hear SIP clients. After some<br>
> troubleshooting we have determined that hte INVITE is sending my<br>
> local(192.168) IP. How would I get * to send the public IP instead<br>
> of the local one? I have changed every IP/domain setting in sip.conf<br>
> to reflect my public IP but it still doesnt want to work. Thanks to<br>
> anyone hthat can help me.<br>
<br>
</div></div>If you show us your config so we can see what is wrong...<br>
<br>
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