I am new to asterisk and need to dig up some info on how to set it all
up. It looks a bit daunting especially all the options available in the
.conf files.
I have 2 SIP phones, GXP2000 and a budgettone 100.
phone1 - 192.168.0.160/24 extension 1000
phone2 - 192.168.0.161/24 extension 1001
Server - 192.168.0.57
I get the following all the time, but can make calls between the 2
extensions, 1000 and 1001 after a long time with forbidden messages on
phones.
My questions are,
1. Do these phones need to register with the server
2. Where does the authentication info go in the SIP.conf & Extensions.conf.
3. Where do I find some good documentation on asterisk/ the conf files.
Apologies for the appearance below.
Aug 27 17:51:03 NOTICE[3877]: chan_sip.c:4045 sip_reg_timeout: --
Registration for 'phone1@192.168.0.57' timed out, trying again
Aug 27 17:51:03 NOTICE[3877]: chan_sip.c:4922 register_verify: Peer
'phone1' is trying to register, but not configured as host=dynamic
Aug 27 17:51:03 NOTICE[3877]: chan_sip.c:7733 handle_request:
Registration from '<sip:phone1@192.168.0.57>' failed for
'192.168.0.57'
Aug 27 17:51:03 WARNING[3877]: chan_sip.c:6869 handle_response:
Forbidden - wrong password on authentication for REGISTER for 'phone1'
to '192.168.0.57'
Aug 27 17:51:10 NOTICE[3877]: chan_sip.c:4045 sip_reg_timeout: --
Registration for 'phone2@192.168.0.57' timed out, trying again
Aug 27 17:51:10 NOTICE[3877]: chan_sip.c:4922 register_verify: Peer
'phone2' is trying to register, but not configured as host=dynamic
Aug 27 17:51:10 NOTICE[3877]: chan_sip.c:7733 handle_request:
Registration from '<sip:phone2@192.168.0.57>' failed for
'192.168.0.57'
Aug 27 17:51:10 WARNING[3877]: chan_sip.c:6869 handle_response:
Forbidden - wrong password on authentication for REGISTER for 'phone2'
to '192.168.0.57'
My sip.conf
=====================[phone1]
username=phone1[root@asterisk asterisk]# cat sip.conf|more
;
; 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@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@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=sip
;context=default ; Default context for incoming calls
;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 ;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 ; 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;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:password@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
externip = a.b.c.d ; 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=192.168.0.0/255.255.255.0; 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
; 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 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@mysipprovider.com
;
; 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 NO
type=friend ; either "friend" (peer+user),
"peer" or
"user"
context=sip
fromuser=phone1 ; overrides the callerid, e.g. required by FWD
callerid="1000" <1000>
secret=1000
;host=192.168.0.160 ; we have a static but private IP address
host=dynamic
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
incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
mailbox=1000@default ; mailbox 1234 in voicemail context "default"
disallow=all ; need to disallow=all before we can use
allowallow=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
; amaflags
; incominglimit
; restrictcid
; mailbox
; username
; template
; fromdomain
; regexten
; fromuser
; host
; mask
; port
; qualify
; defaultip
; rtptimeout
; rtpholdtimeout
;[sip_proxy]
; For incoming calls only. Example: FWD (Free World Dialup)
;type=user
;context=from-fwd
;[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!
;host=box.provider.com
;[grandstream1]
;type=friend ; either "friend" (peer+user),
"peer" or
"user"
;context=from-sip
;fromuser=grandstream1 ; overrides the callerid, e.g. required
by FWD
;callerid=John Doe <1234>
;host=192.168.0.23 ; we have a static but private IP address
;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
;incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
;mailbox=1234@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
[phone2]
username=phone2
type=friend ; either "friend" (peer+user),
"peer" or
"user"
context=sip
secret=1001
fromuser=phone2 ; overrides the callerid, e.g. required by FWD
callerid="1001" <1001>
host=dynamic
;host=192.168.0.161 ; we have a static but private IP address
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
incominglimit=1 ; permit only 1 outgoing call at a time
; from the phone to asterisk
mailbox=1001@default ; mailbox 1234 in voicemail context
"default"
disallow=all ; need to disallow=all before we can use
allowallow=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
===================
My Extensions.conf
===================
[root@asterisk asterisk]# cat extensions.conf|more
[sip]
exten => 1000,1,Dial(SIP/phone1,20,tr)
exten => 1001,1,Dial(SIP/phone2,20,tr)
exten => 1002,1,Dial(SIP/phone1&SIP/phone2,20,tr)
rest is as per extensions.conf.sample except commenting out the section
at the bottom referring to extension 1000.
===================
Thanks
--
Gary
