hi every1
i have a little problem (a big problem to me)
yes, another newbie guy
hardware:
a linksys spa841 (SIP voip phone)
a router texas instruments (4 ports)
a pc(amd64 athlon,512 mb)
OS ubuntu feysti 7.04 (x86)
asterisk 1.14.18
Description:
I can’t register sip phone to mi asterisk
*router -->> 192.168.1.1
*ip pc -->> 192.168.1.33
*ip linksys spa 841 -->> 192.168.1.129
my configuration files (just like www.asteriskdocs.org)
###@@@@@@@@@@@@@@@###
extensions.conf
###@@@@@@@@@@@@@@@###
[globals]
[general]
clearglobalvars=yes
[default]
exten => s,1,Verbose(1|Unrouted call handler)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,Playback(tt-weasels)
exten => s,n,Hangup()
[incoming_calls]
[internal]
exten => 500,1,Verbose(1|Echo test application)
exten => 500,n,Echo()
exten => 500,n,Hangup()
[phones]
include => internal
###@@@@@@@@@@@@@@@###
sip.conf
###@@@@@@@@@@@@@@@###
[general]
context=default ; Default context for incoming calls
allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will
; listen on
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
[authentication]
[1000]
type=friend
context=phones
host=192.168.1.129
====================================================
- this is how to registers my phone *
====================================================
- General → Line Enable → yes
- NAT Settings → NAT Mapping Enable → no
- NAT Settings → NAT Keep Alive Enable → no
- Proxy and Registration → Proxy → 192.168.1.33
- Proxy and Registration → Register → yes
- Proxy and Registration → Make Call Without Reg → no
- Proxy and Registration → Ans Call Without Reg → no
- Subscriber Information → Display Name → Caller ID information
- Subscriber Information → User ID → 1000
- Subscriber Information → Password →
- Subscriber Information → Use Auth ID → yes
- Subscriber Information → Auth ID → 1000
- Audio Configuration → Preferred Codec → G711u
- Audio Configuration → Use Pref Codec Only → no
- Audio Configuration → Silence Supp Enable → no
- Audio Configuration → DTMF Tx Method → Auto
- Submit All Changes
i dont tone on the phone (any ligth turn green, i just see the fucking orange ligth & i hate it)
Thanks for the answer. I really appreciated
bye