Sip voip phone problem with asterisk - no registry

Asterisk Asterisk Support
#1

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 *
    ====================================================
  1. General → Line Enable → yes
  2. NAT Settings → NAT Mapping Enable → no
  3. NAT Settings → NAT Keep Alive Enable → no
  4. Proxy and Registration → Proxy → 192.168.1.33
  5. Proxy and Registration → Register → yes
  6. Proxy and Registration → Make Call Without Reg → no
  7. Proxy and Registration → Ans Call Without Reg → no
  8. Subscriber Information → Display Name → Caller ID information
  9. Subscriber Information → User ID → 1000
  10. Subscriber Information → Password →
  11. Subscriber Information → Use Auth ID → yes
  12. Subscriber Information → Auth ID → 1000
  13. Audio Configuration → Preferred Codec → G711u
  14. Audio Configuration → Use Pref Codec Only → no
  15. Audio Configuration → Silence Supp Enable → no
  16. Audio Configuration → DTMF Tx Method → Auto
  17. 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

#2

sip.conf should include:

[1000]
type=friend
context=phones
host=192.168.1.129
username=1000
secret=1000

username and secret of your choice.
Put the same settings in the phone.

BTW. swearing does not help.

#3

hi gazmac52

I do that and doesn’t work either

But something really rare happens after this post my asterisk server and the linksys phone start to work. I make a call record adding:

exten => 100,1,Wait(2)
exten => 100,n,Record(/var/lib/asterisk/sounds/recording:gsm)
exten => 100,n,NoOp(${RECORDED_FILE})
exten => 100,n,Wait(2)
exten => 100,n,Playback(/var/lib/asterisk/sounds/recording)
exten => 100,n,NoOp(${PLAYBACKSTATUS})
exten => 100,n,Wait(1)
exten => 100,n,Hangup()

to extensions.conf file([internal] section) and that was good, but after reboot i have the same problem again. i really don’t understand. I hope some1 give more ideas.

thanks

#4

i almost forget

this is the error message

[Mar 11 23:22:04] ERROR[11030]: chan_sip.c:8583 register_verify: Peer ‘1000’ is trying to register, but not configured as host=dynamic
[Mar 11 23:22:07] NOTICE[11030]: chan_sip.c:15069 handle_request_register: Registration from ‘“1000” sip:1000@192.168.1.33’ failed for ‘192.168.1.129’ - Peer is not supposed to register

#5

Here is a working example, change it to suit your config.

[202]
type=friend
context=internal
username=202
callerid=Conference Rm <202>
host=dynamic
secret=202
nat=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g729
amaflags=default
accountcode=20
pickupgroup=1
call-limit=1

I expect SPA841 is similar to SPA942 which I have.
If so, set:
9. Subscriber Information → User ID → 1000
10. Subscriber Information → Password → 1000 or pw as in sip.conf
11. Subscriber Information → Use Auth ID → no
12. Subscriber Information → Auth ID → blank

#6

Hi gazmac52

Your example works for me too.
You’re right about the Use Auth ID -> no

thank you very much!
muchas gracias.