hello,
I try to set up two lines per ip 300 phone,
registration is ok but i get Failure to authenticate
407 for subscribe.
Anybody could help me to configure Asterisk in order
to set instant message and presence ?
I've tried with Ondo sip server it's ok !
Regards
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Cr�ez votre Yahoo! Mail sur http://fr.mail.yahoo.com/;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no
; You can include other config files, use the #include command (without the ';')
; Note that this is different from the "include" command that includes contexts
within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"
; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
[sip]
exten => 100,1,Dial(SIP/100)
exten => 150,1,Dial(SIP/150)
exten => 200,1,Dial(SIP/200)
exten => 200,1,Dial(SIP/250)
;
; 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/[EMAIL PROTECTED] 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/[EMAIL PROTECTED]
; 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=sip ; Default context for incoming calls
;recordhistory=yes ; Record SIP history by default
; (see sip history / sip no history)
realm=home.net ; 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=192.168.0.2 ; IP address to bind to (0.0.0.0 binds to all)
[100]
type=friend
username=100
secret=100
fromuser=100
host=dynamic
context=sip
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
"never"
[150]
type=friend
username=150
secret=150
fromuser=150
host=dynamic
context=sip
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
[200]
type=friend
username=200
fromuser=200
secret=200
host=dynamic
context=sip
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
"never"
[250]
type=friend
username=250
fromuser=250
secret=250
host=dynamic
context=sip
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
progressinband=no ; Polycom phones don't work properly with
"never"
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