Ich hab eigentlich die Einstellung in der Sip gemacht:
[general]
context=default ; Default context for incoming calls
; port 5060 already used by fritzbox, so use 5061
; and reconfigure local sip phones (remote sip goes through asterisk!)
; ^^^^^
; XLITE in same subnet example config (fritzbox 192.168.222.100)
; - System Settings / SIP Proxy / Default
; - Enabled: Yes
; - Display Name=Username=Auth. User=: 771
; - Password: 771
; - Domain/Realm: 192.168.222.100
; - SIP Proxy: 192.168.222.100:5061
; ^^^^
; - Outbound Proxy: (empty)
; - Use Outbound Proxy: Never
; - Send Internal IP: Always
; ^^^^^^
; - Register: Default
; - Advanced System Settings / Audio / Silence Settings
; - Transmit Silence: Yes
bindport=5061 ; 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
language=de
;using external sip provider
;and coping with dynamic ip address (the entry localnet=127...
;is required if the avm sip client is registered at localhost:5061)
;(idea taken from
[email protected])
externhost=testkc.dyndns.org
;localnet=127.0.0.0/255.0.0.0
nat=yes
externrefresh=10
canreinvite=no
localnet=192.168.3.0/255.255.255.0
;register => bluesip/username
[email protected]/sip1
;...(
http://www.ip-phone-forum.de/showpost.php?p=500468&postcount=12)
[771]
context=sip771
callerid="TestSIP 771" <771>
host=dynamic
domain=192.168.3.1
nat=yes
qualify=no ; X-Lite is behind a NAT router
type=friend
user=771
secret=xxx
;mailbox=771
canreinvite=no ; Typically set to NO if behind NAT
;regexten=1234 ; When they register, create extension 1234
;username=xlite1
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
Hier die ar7.cfg:
mcupstream = "internet";
voip_forwardrules = "udp 0.0.0.0:5060 0.0.0.0:5060",
"tcp 0.0.0.0:5060 0.0.0.0:5060",
"udp 0.0.0.0:5061 0.0.0.0:5061",
"tcp 0.0.0.0:5061 0.0.0.0:5061",
"udp 0.0.0.0:7078+32 0.0.0.0:7078";
Danke
Thommy