[Asterisk-Users] polycom soundpoint ip 300

harry gaillac
Wed, 09 Feb 2005 08:10:33 -0000

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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-------------- next part --------------
;
; 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)

-------------- next part --------------
;
; 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
;

[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"