[Asterisk-Users] Set up SIP,
now I'm getting a busy tone and weird (to me)messages on the console
beonice
beonice at yahoo.com
Sun May 1 19:12:43 MST 2005
--- Tim Connolly <tim at timsnet.com> wrote:
> Is NAT=yes on, are you behind a firewall? Give us
> some connectivity details.
> Usually when you see maximum retries, its because
> you have one-way
> communications with the far end for some reason. Are
> you setting "externip"
> statically?
To answer your questions, yes, I am behind a firewall.
The asterisk server is the only device connected to a
cheapo Netgear 4-port router/firewall. I'm not setting
externip myself, so whatever the default is, it's
getting used. I'm also NOT making outgoing calls, and
there are no actual SIP devices attached ... I'm just
trying to receive incoming calls forwarded from a
different provider via SIP.
Here is a complete sip.conf file ... do I need to
provide anything else?
sip.conf:
;
; 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 at 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 at 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
;
[general]
;context=default ; Default
context for incoming calls
context=unwelcome-calls ; Default context for
incoming calls
; After all, we don't
want any random
; incoming calls to
have access to outbound
; calling
;recordhistory=yes ; Record SIP history
by default
; (see sip history /
sip no history)
;realm=mydomain.tld ; 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
port=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
;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
;maxexpirey=3600 ; Max length of
incoming registration we allow
;defaultexpirey=120 ; Default length of
incoming/outoing registration
;notifymimetype=text/plain ; Allow overriding of
mime type in MWI NOTIFY
;videosupport=yes ; Turn on support for
SIP video
;disallow=all ; First disallow all
codecs
;allow=ulaw ; Allow codecs in
order of preference
;allow=ilbc ; Note: codec order is
respected only in [general]
;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
;progressinband=no ; If we should
generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change
the user agent string
;nat=no ; NAT settings
; 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)
;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.
;
; 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=iaxregistrations
;
; 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
;
; This will pass incoming calls to the 's'
extension
;
;
;register => 2345:password at sip_proxy/1234
;
; Register 2345 at sip provider 'sip_proxy'. Calls
from this provider connect to local
; extension 1234 in extensions.conf default
context, unless you define
; 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
; register for SIP at VoicePulseConnect
register => uid:secret at srvr.voicepulse.com
;externip = 200.201.202.203 ; 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
; 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
;-----------------------------------------------------------------------------------
; Users and peers have different settings available.
Friends have all settings,
; since a friend is both a peer and a user
;
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; permit permit
; deny deny
; auth auth
; secret secret
; md5secret md5secret
; dtmfmode dtmfmode
; canreinvite canreinvite
; nat nat
; callgroup callgroup
; pickupgroup pickupgroup
; language language
; allow allow
; disallow disallow
; insecure insecure
; trustrpid trustrpid
; progressinband progressinband
; promiscredir promiscredir
; callerid
; accountcode
; amaflags
; incominglimit
; restrictcid
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
[888] ; For incoming calls ONLY
type=user ; This device takes incoming calls
username=uid ; Username on device
secret=secret ; Password for device
host=srvr.voicepulse.com ; This host doesn't change
frequently
context=allowed_context ; Inbound calls from this
; host go to the normal
; context
...
at the end of this sip.conf are a bunch of commented
out details for particular SIP phones. As I have no
SIP phones, I've left all of these commented out.
Thanks,
Maya
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