Problem registring sip phone in realtime

Asterisk Asterisk Support
1

Im triying to go realtime. But i cant register my sip phones. I alway get the error :

Aug 11 17:34:29 NOTICE[3394] chan_sip.c: Registration from '220 <sip:220@192.168.53.5:5060>' failed for '192.168.53.53' - Username/auth name mismatch

But i don’t understand … I have the same information in my sip table that i had in my sip.conf :
Sip.conf :

;[220]
;type=friend
;context=torrent-out
;username=220@torrent
;secret=****
;dtmfmode=rfc2833
;mailbox=220@torrent
;nat=yes
;host=64.26.157.252
;canreinvite=yes
;qualify=yes
;host=dynamic
;disallow=all
;allow=ulaw
;allow=gsm
;allow=alaw

Sip table :

name : 220
context:torrent-out
dtmfmode: rfc2833
host:dynamic
qualify:yes
canreinvite=yes
mailbox:220@torrent  	
secret: ****
type: friend
 username:220@torrent

My phone was working well before i transfert de configuration on the table. I can’t find what i did wrong !

2

Anyone ??? Im really short on idea here.

3

You have specified username as 220@torrent.
Make it the same as the context i.e 220 and then see if it works.

4

tried it … does’nt work :frowning:

5

5$ paypal to anyone that help me find the solution :stuck_out_tongue:

6

Here is my ngrep -W byline port 5060 result about this phone

U 192.168.53.110:5060 -> 192.168.53.5:5060
REGISTER sip:192.168.53.5 SIP/2.0.
Via: SIP/2.0/UDP 192.168.53.110;branch=z9hG4bK80f26f544e2adb11839300096b5ac280.
From: 220 <sip:220@192.168.53.5>;tag=8223.
To: <sip:220@192.168.53.5>.
Call-ID: 80C53E53-4E2A-DB11-8393-00096B5AC280@192.168.53.110.
CSeq: 1 REGISTER.
Contact: <sip:220@192.168.53.110>.
Max-Forwards: 70.
User-Agent: SIPPER for PhonerLite.
Expires: 900.
Content-Length: 0.
.

#
U 192.168.53.5:5060 -> 192.168.53.110:5060
SIP/2.0 404 Not found.
Via: SIP/2.0/UDP 192.168.53.110;branch=z9hG4bK80f26f544e2adb11839300096b5ac280;received=192.168.53.110.
From: 220 <sip:220@192.168.53.5>;tag=8223.
To: <sip:220@192.168.53.5>;tag=as2f2142ae.
Call-ID: 80C53E53-4E2A-DB11-8393-00096B5AC280@192.168.53.110.
CSeq: 1 REGISTER.
User-Agent: Asterisk PBX.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Contact: <sip:220@192.168.53.5>.
Content-Length: 0.

I see that there is a 404 not found error but i don’t understand why ?

7

sip.conf entries above appear to be commented out?

8

well i know its because im going into realtime this information is now in a mysql database