steve.langstaff at cit...
Guest
|
 Posted: Mon Jan 21, 2008 7:29 am Post subject: [asterisk-users] Here is my sip.conf I am having a problemco |
 |
|
It looks like you are attempting to register 'extensions' 6000 and 1000.
You need to define these in sip.conf (infact, the sip.conf you provided appears to have no edits at all).
| Quote: | -----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Andrew Ladanowski
Sent: 21 January 2008 11:55
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Here is my sip.conf I am having
a problemconnecting my X-Litetomy Asterix box
I keep getting Registration error 404-Not found When I look
at the log file I get.
[Jan 20 14:17:46] NOTICE[2637] chan_sip.c: Registration from
'<sip:6000 at 192.168.3.125@192.168.3.125>' failed for
'192.168.3.116' - Device does not match ACL [Jan 20 14:21:25]
NOTICE[2637] chan_sip.c: Registration from
'<sip:6000 at 192.168.3.125@192.168.3.125>' failed for
'192.168.3.116' - Device does not match ACL [Jan 20 14:22:47]
NOTICE[2637] chan_sip.c: Registration from
'"Csilla"<sip:1000 at 192.168.3.128>' failed for '192.168.3.116'
- Device does not match ACL [Jan 20 14:25:09] NOTICE[2637]
chan_sip.c: Registration from
'<sip:6000 at 192.168.3.125@192.168.3.125>' failed for
'192.168.3.116' - Device does not match ACL [Jan 20 14:28:47]
NOTICE[2637] chan_sip.c: Registration from
'<sip:6000 at 192.168.3.125@192.168.3.125>' failed for
'192.168.3.116' - Device does not match ACL [] Here is my
sip.conf I am having a problemconnecting my X-Litetomy Asterix box
; SIP Configuration example 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
;
; reload chan_sip.so Reload configuration file
; Active SIP peers will not be
reconfigured
;
[general]
context=default ; Default context for
incoming calls
;allowguest=no ; Allow or reject guest
calls (default is yes)
allowoverlap=no ; Disable overlap
dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless
enabled in peers or users)
; Default is enabled
;realm=mydomain.tld ; Realm for digest authentication
; 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
;domain=mydomain.tld ; 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
;pedantic=yes ; Enable checking of tags in headers,
; international character
conversions in URIs
; and multiline formatted
headers for strict
; SIP compatibility (defaults to "no")
; See doc/README.tos for a description of these parameters.
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time
of incoming registrations
; and subscriptions (seconds)
;minexpiry=60 ; Minimum length of
registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of
incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for
messages to monitored hosts
; Defaults to 100 ms
;notifymimetype=text/plain ; Allow overriding of mime type
in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox
checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't
support the MWI RFC
; fully. Enable this option to
not get error messages
; when sending MWI to phones
with this bug.
;vmexten=voicemail ; dialplan extension to reach
mailbox sets the
; Message-Account in the MWI
notify message
; defaults to "asterisk"
;disallow=all ; First disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
;allow=ilbc ; see doc/rtp-packetization for
framing options
;
; This option specifies a preference for which music on hold
class this channel ; should listen to when put on hold if the
music class has not been set on the ; channel with
Set(CHANNEL(musicclass)=whatever) in the dialplan, and the
peer ; channel putting this one on hold did not suggest a music class.
;
; This option may be specified globally, or on a per-user or
per-peer basis.
;
;mohinterpret=default
;
; This option specifies which music on hold class to suggest
to the peer channel ; when this channel places the peer on
hold. It may be specified globally or on ; a per-user or
per-peer basis.
;
;mohsuggest=default
;
;language=en ; Default language setting for
all users/peers
; This may also be set for
individual users/peers
;relaxdtmf=yes ; Relax dtmf handling
;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 Asterisk 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.
;
;videosupport=yes ; Turn on support for SIP
video. You need to turn this on
; 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.
;maxcallbitrate=384 ; Maximum bitrate for video
calls (default 384 kb/s)
; Videosupport and
maxcallbitrate is settable
; for peers and users as well
;callevents=no ; generate manager
events when sip ua
; performs events (e.g. hold)
;alwaysauthreject = yes ; When an incoming
INVITE or REGISTER is to be rejected,
; 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
;g726nonstandard = yes ; If the peer
negotiates G726-32 audio, use AAL2 packing
; 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 
;matchexterniplocally = yes ; Only substitute the
externip or externhost setting if it matches
; your localnet setting.
Unless you have some sort of strange network
; setup you will not need to
enable this.
;
; 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 and have a
"regexten=" configuration item.
; Multiple contexts may be specified by separating them with
'&'. The ; actual extension is the 'regexten' parameter of
the registering peer or its ; name if 'regexten' is not
provided. If more than one context is provided, ; the
context must be specified within regexten by appending the
desired ; context after '@'. More than one regexten may be
supplied if they are ; separated by '&'. Patterns may be
used in regexten.
;
;regcontext=sipregistrations
;
;--------------------------- RTP timers
----------------------------------------------------
; These timers are currently used for both audio and video
streams. The RTP timeouts ; are only applied to the audio channel.
; The settings are settable in the global section as well as
per device ;
;rtptimeout=60 ; Terminate call if 60
seconds of no RTP or RTCP activity
; 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.
;rtpholdtimeout=300 ; Terminate call if 300 seconds
of no RTP or RTCP activity
; on the audio channel
; when we're on hold (must be >
rtptimeout)
;rtpkeepalive=<secs> ; Send keepalives in the RTP
stream to keep NAT open
; (default is off - zero)
;--------------------------- SIP DEBUGGING
---------------------------------------------------
;sipdebug = yes ; Turn on SIP debugging
by default, from
; the moment the channel loads
this configuration
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
;dumphistory=yes ; Dump SIP history at end of
SIP dialogue
; SIP history is output to the
DEBUG logging channel
;--------------------------- STATUS NOTIFICATIONS
(SUBSCRIPTIONS) ---------------------------- ; You can
subscribe to the status of extensions with a "hint" priority
; (See extensions.conf.sample for examples) ; chan_sip
support two major formats for notifications: dialog-info and
SIMPLE ; ; You will get more detailed reports (busy etc) if
you have a call limit set ; for a device. When the call limit
is filled, we will indicate busy. Note that ; you need at
least 2 in order to be able to do attended transfers.
;
; For queues, you will need this level of detail in status
reporting, regardless ; if you use SIP subscriptions. Queues
and manager use the same internal interface ; for reading
status information.
;
; Note: Subscriptions does not work if you have a realtime
dialplan and use the ; realtime switch.
;
;allowsubscribe=no ; Disable support for
subscriptions. (Default is yes)
;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 (default: no)
;notifyhold = yes ; Notify subscriptions on HOLD
state (default: no)
; Turning on notifyringing and
notifyhold will add a lot
; more database transactions if
you are using realtime.
;limitonpeers = yes ; Apply call limits on peers
only. This will improve
; 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.
;----------------------------------------- T.38 FAX
PASSTHROUGH SUPPORT ----------------------- ; ; This setting
is available in the [general] section as well as in device
configurations.
; Setting this to yes, enables T.38 fax (UDPTL) passthrough
on SIP to SIP calls, provided ; both parties have T38 support
enabled in their Asterisk configuration ; This has to be
enabled in the general section for all devices to work. You
can then ; disable it on a per device basis.
;
; T.38 faxing only works in SIP to SIP calls, with no local
or agent channel being used.
;
; t38pt_udptl = yes ; Default false
;
;----------------------------------------- OUTBOUND SIP
REGISTRATIONS ------------------------ ; 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 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
;----------------------------------------- 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 = 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
;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=no ; Global NAT settings
(Affects all peers and users)
; 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)
;----------------------------------- MEDIA HANDLING
-------------------------------- ; By default, Asterisk tries
to re-invite the audio to an optimal path. If there's ; no
reason for Asterisk to stay in the media path, the media will
be redirected.
; This does not really work with in the case where Asterisk
is outside and have ; clients on the inside of a NAT. In that
case, you want to set canreinvite=nonat ;
;canreinvite=yes ; Asterisk by default tries to
redirect the
; 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).
;directrtpsetup=yes ; Enable the new experimental
direct RTP setup. This sets up
; 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 INVITE.
;canreinvite=nonat ; An additional option is to
allow media path redirection
; (reinvite) but only when the
peer where the media is being
; sent is known to not be
behind a NAT (as the RTP core can
; determine it based on the
apparent IP address the media
; arrives from).
;canreinvite=update ; Yet a third option... use
UPDATE for media path redirection,
; instead of INVITE. This can
be combined with 'nonat', as
; 'canreinvite=update,nonat'.
It implies 'yes'.
;----------------------------------------- REALTIME SUPPORT
------------------------ ; For additional information on ARA,
the Asterisk Realtime Architecture, ; please read
realtime.txt and extconfig.txt in the /doc directory of the ;
source code.
;
;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)
;rtsavesysname=yes ; Save systemname in realtime
database at registration
; Default= 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
;----------------------------------------- SIP DOMAIN SUPPORT
------------------------ ; 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
;domain=mydomain.tld,mydomain-incoming
; 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=no ; 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.
; fromdomain=mydomain.tld ; 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.
;------------------------------ JITTER BUFFER CONFIGURATION
--------------------------
; jbenable = yes ; Enables the use of a
jitterbuffer on the receiving side of a
; SIP channel. Defaults to
"no". An enabled jitterbuffer will
; be used only if the sending
side can create and the receiving
; side can not accept jitter.
The SIP channel can accept jitter,
; thus a jitterbuffer on the
receive SIP side will be used only
; if it is forced and enabled.
; jbforce = no ; Forces the use of a
jitterbuffer on the receive side of a SIP
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the
jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps
over which the jitterbuffer is
; resynchronized. Useful to
improve the quality of the voice, with
; big jumps in/broken
timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation,
used on the receiving side of a SIP
; channel. Two implementations
are currently available - "fixed"
; (with size always equals to
jbmaxsize) and "adaptive" (with
; variable size, actually the
new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame
logging. Defaults to "no".
;-------------------------------------------------------------
----------------------
[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:topsecret at digium.com
;
; You may also add auth= statements to [peer] definitions ;
Peer auth= override all other authentication settings if we
match on realm
;-------------------------------------------------------------
-----------------
; 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
; callingpres callingpres
; permit permit
; deny deny
; 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
; useclientcode useclientcode
; accountcode accountcode
; setvar setvar
; callerid callerid
; amaflags amaflags
; call-limit call-limit
; allowoverlap allowoverlap
; allowsubscribe allowsubscribe
; allowtransfer allowtransfer
; subscribecontext subscribecontext
; videosupport videosupport
; maxcallbitrate maxcallbitrate
; rfc2833compensate mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
; sendrpid
; outboundproxy
; rfc2833compensate
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup) ;
We match on IP address of the proxy for incoming calls ;
since we can not match on username (caller id) ;type=peer
;context=from-fwd ;host=fwd.pulver.com
;[sip_proxy-out]
;type=peer ; we only want to call
out, not be called
;secret=guessit
;username=yourusername ;
Authentication user for outbound proxies
;fromuser=yourusername ; Many SIP
providers require this!
;fromdomain=provider.sip.domain
;host=box.provider.com
;usereqphone=yes ; This provider
requires ";user=phone" on URI
;call-limit=5 ; permit only 5
simultaneous outgoing calls to this peer
;outboundproxy=proxy.provider.domain ; send outbound
signaling to this proxy, not directly to the peer
; Call-limits will not
be enforced on real-time peers,
; since they are not
stored in-memory
;port=80 ; The port number we
want to connect to on the remote side
; Also used as
"defaultport" in combination with "defaultip" settings
;-------------------------------------------------------------
-----------------
; Definitions of locally connected SIP devices ;
; type = user a device that authenticates to us by "from"
field to place calls
; type = peer a device we place calls to or that calls us and
we match by host
; type = friend two configurations (peer+user) in one ; ; For
device names, we recommend using only a-z, numerics (0-9) and
underscore ; ; For local phones, type=friend works most of
the time ; ; If you have one-way audio, you probably have NAT
problems.
; If Asterisk is on a public IP, and the phone is inside of a
NAT device ; you will need to configure nat option for those phones.
; Also, turn on qualify=yes to keep the nat session open
;[grandstream1]
;type=friend
;context=from-sip ; Where to start in the
dialplan when this phone calls
;callerid=John Doe <1234> ; Full caller ID, to override
the phones config
; on incoming calls to Asterisk
;host=192.168.0.23 ; we have a static but private
IP address
; No registration allowed
;nat=no ; there is not NAT
between phone and Asterisk
;canreinvite=yes ; allow RTP voice traffic to
bypass Asterisk
;dtmfmode=info ; either RFC2833 or
INFO for the BudgeTone
;call-limit=1 ; permit only 1 outgoing call
and 1 incoming call at a time
; from the phone to asterisk
; 1 for the explicit peer, 1
for the explicit user,
; remember that a friend equals
1 peer and 1 user in
; memory
; This will affect your
subscriptions as well.
; There is no combined call
counter for a "friend"
; so there's currently no way
in sip.conf to limit
; to one inbound or outbound
call per phone. Use
; the group counters in the
dial plan for that.
;
;mailbox=1234 at default ; mailbox 1234 in voicemail
context "default"
;disallow=all ; need to disallow=all before
we can use allow=
;allow=ulaw ; Note: In user sections the
order of codecs
; listed with allow= does NOT matter!
;allow=alaw
;allow=g723.1 ; Asterisk only supports g723.1
pass-thru!
;allow=g729 ; Pass-thru only unless g729
license obtained
;callingpres=allowed_passed_screen ; Set caller ID presentation
; See README.callingpres for
more information
;[xlite1]
; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
; Note that Xlite sends NAT keep-alive packets, so
qualify=yes is not needed ;type=friend
;regexten=1234 ; When they register,
create extension 1234
;callerid="Jane Smith" <5678>
;host=dynamic ; This device needs to register
;nat=yes ; X-Lite is behind a NAT router
;canreinvite=no ; Typically set to NO
if behind NAT
;disallow=all
;allow=gsm ; GSM consumes far less
bandwidth than ulaw
;allow=ulaw
;allow=alaw
;mailbox=1234 at default,1233 at default ; Subscribe to status
of multiple mailboxes
;[snom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls
from this user
;secret=blah
;subscribecontext=localextensions ; Only allow SUBSCRIBE
for local extensions
;language=de ; Use German prompts for this user
;host=dynamic ; This peer register with us
;dtmfmode=inband ; Choices are inband, rfc2833, or info
;defaultip=192.168.0.59 ; IP used until peer registers
;mailbox=1234 at context,2345 ; Mailbox(-es) for message
waiting indicator
;subscribemwi=yes ; Only send notifications if this phone
; subscribes for mailbox notification
;vmexten=voicemail ; dialplan extension to reach mailbox
; sets the Message-Account in
the MWI notify message
; defaults to global vmexten
which defaults to "asterisk"
;disallow=all
;allow=ulaw ; dtmfmode=inband only works
with ulaw or alaw!
;[polycom]
;type=friend ; Friends place calls and receive calls
;context=from-sip ; Context for incoming calls
from this user
;secret=blahpoly
;host=dynamic ; This peer register with us
;dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
;username=polly ; Username to use in
INVITE until peer registers
; Normally you do NOT need to
set this parameter ;disallow=all
;allow=ulaw ; dtmfmode=inband only works
with ulaw or alaw!
;progressinband=no ; Polycom phones don't work
properly with "never"
;[pingtel]
;type=friend
;secret=blah
;host=dynamic
;insecure=port ; Allow matching of
peer by IP address without
; matching port number
;insecure=invite ; Do not require authentication
of incoming INVITEs
;insecure=port,invite ; (both)
;qualify=1000 ; Consider it down if it's 1
second to reply
; Helps with NAT session
; qualify=yes uses default value
;
; Call group and Pickup group should be in the range from 0 to 63 ;
;callgroup=1,3-4 ; We are in caller groups 1,3,4
;pickupgroup=1,3-5 ; We can do call pick-p for
call group 1,3,4,5
;defaultip=192.168.0.60 ; IP address to use if
peer has not registered
;deny=0.0.0.0/0.0.0.0 ; ACL: Control access to this
account based on IP address
;permit=192.168.0.60/255.255.255.0
;[cisco1]
;type=friend
;secret=blah
;qualify=200 ; Qualify peer is no more than
200ms away
;nat=yes ; This phone may be natted
; Send SIP and RTP to the IP
address that packet is
; received from instead of
trusting SIP headers
;host=dynamic ; This device registers with us
;canreinvite=no ; Asterisk by default
tries to redirect the
; 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).
;defaultip=192.168.0.4 ; IP address to use
until registration
;username=goran ; Username to use when
calling this device before registration
; Normally you do NOT need to
set this parameter
;setvar=CUSTID=5678 ; Channel variable to be set
for all calls from this device
;[pre14-asterisk]
;type=friend
;secret=digium
;host=dynamic
;rfc2833compensate=yes ; Compensate for
pre-1.4 DTMF transmission from another Asterisk machine.
; You must have this turned on
or DTMF reception will work improperly.
Andrew Ladanowski
AddInSolutions Inc.
www.addinsol.com
andrew at addinsol.com
Phone: 954-815-2402
Fax: 954-414-8432
?
?
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-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Shane D
Sent: Sunday, January 20, 2008 9:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] I am having a problem
connecting my X-Litetomy Asterix box
Basically, You will need to send the sip.conf file. It will
not work unless you have stuff set up in sip.conf.
x-Lite works fine; I'm using it without a hitch.
HTH,
Shane
On 1/20/08, Erik Anderson <erikerik at gmail.com> wrote:
| Quote: | On Jan 20, 2008 8:06 PM, Andrew Ladanowski
<Andrew at addinsolutionsinc.com>
wrote:
Andrew - you're going to need to get us your sip.conf before we can
really assist you any further.
_______________________________________________
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--
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712
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To UNSUBSCRIBE or update options visit:
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_______________________________________________
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