I am having trouble setting this system up and wonder if some one help me.
Does anyone know what is missing if anything to get 2 phones on my
asterisk home server to be able to call each other.
I have a WRT54GS running OpenWRT/asterisk connected to a PAP2 with 2
extensions 5060/5061, this is on the lan side of my gateway/router
WRT54G 192.168.1.1
BusyBox v1.00 (2006.11.07-01:40+0000) Built-in shell (ash)
Asterisk 1.2.1, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <[EMAIL PROTECTED]>
=========================================================================
Connected to Asterisk 1.2.1 currently running on OpenWrt (pid = 5084)
OpenWrt*CLI> sip show settings
Global Settings:
----------------
SIP Port: 5060
Bindaddress: 0.0.0.0
Videosupport: No
AutoCreatePeer: No
Allow unknown access: Yes
Promsic. redir: No
SIP domain support: No
Call to non-local dom.: Yes
URI user is phone no: No
Our auth realm asterisk
Realm. auth: No
User Agent: Asterisk PBX
MWI checking interval: 10 secs
Reg. context: (not set)
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
IP ToS: 0x0
OSP Support: No
SIP realtime: Disabled
Global Signalling Settings:
---------------------------
Codecs: none
Relax DTMF: No
Compact SIP headers: No
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: No
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Default Settings:
-----------------
Context: default
Nat: RFC3581
DTMF: rfc2833
Qualify: 0
Use ClientCode: No
Progress inband: Never
Language: (Defaults to English)
Musicclass: default
Voice Mail Extension: asterisk
******************sip.conf file*************************
GNU nano 1.3.8 File: sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=yes ; Allow or reject guest calls (default
is yes, this can also be set to 'osp'
; if asterisk was compiled with OSP support.
;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
bindport=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
;domain=OpenWrt ; 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
;domain=mydomain.tld,mydomain-incoming
; Add domain and configure incoming context
; for external calls to this domain
;domain=192.168.1.130 ; Add IP address as local domain
;domain=192.168.1.135 ; 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.
;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
;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 = auto ; 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
GNU nano 1.3.8 File: sip.conf
; (provider).
;
; host is either a host name defined in DNS or the name of a section defined
; below.
;
; Examples:
;
;register => 1234:[EMAIL PROTECTED]
;
; This will pass incoming calls to the 's' extension
;
;
;register => 2345:[EMAIL PROTECTED]/1234
;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 = 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
;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
;[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
defaultip=192.168.1.130 ; IP address to use until registration
[phone1]
type=friend
context=default
secret=pascal
;host=linksysPAP
host=192.168.1.130
;defaultip=192.168.1.135
username=phone1
dtmfmode=inband ; Choices are inband, rfc2833, or info
mailbox=5560 ; Mailbox for message waiting indicator
callerid="phone1" <5560>
disallow=all
allow=ulaw
[phone2]
type=friend
context=default
secret=pascal
;host=linksysPAP
host=192.168.1.130
;defaultip=192.168.1.135
username=phone2
dtmfmode=inband ; Choices are inband, rfc2833, or info
mailbox=5561 ; Mailbox for message waiting indicator
callerid="phone2" <5561>
disallow=all
allow=ulaw
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
