Asterisk problem with OpenBTS

I’m using ubuntu 10.04
when I run the Asterisk through the terminal I got

[Mar 26 00:02:38] NOTICE[3826]: chan_sip.c:11629 sip_reg_timeout: -- Registration for 'kestrel0@sip.ca2.link2voip.com' timed out, trying again (Attempt #82) [Mar 26 00:02:38] NOTICE[3826]: chan_sip.c:11629 sip_reg_timeout: -- Registration for 'kestrel0@sip.ca1.link2voip.com' timed out, trying again (Attempt #82) [Mar 26 00:02:39] NOTICE[3826]: chan_sip.c:18180 handle_response_register: Failed to authenticate on REGISTER to 'kestrel0@sip.ca2.link2voip.com' (Tries 3) [Mar 26 00:02:40] NOTICE[3826]: chan_sip.c:18180 handle_response_register: Failed to authenticate on REGISTER to 'kestrel0@sip.ca1.link2voip.com' (Tries 3)

this the extensions.conf … that’s what I added to it

[code]#include “extensions.local.conf”
[macro-dialGSM]
exten => s,1,Dial(SIP/${ARG1})
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-CANCEL,1,Hangup
exten => s-NOANSWER,1,Hangup
exten => s-BUSY,1,Busy(30)
exten => s-CONGESTION,1,Congestion(30)
exten => s-CHANUNAVAIL,1,playback(ss-noservice)
exten => s-CANCEL,1,Hangup

[sip-external]

; check for local extensions first
include => sip-local

; other Asterisk server(s)
;;exten => _611,1,Dial(SIP/5078322417@vgw1.link2voip.com)
;;exten => _9XXXX,1,Dial(SIP/${EXTEN:-4}@192.168.0.15)
;exten => 8099,1,Dial(SIP/8099@65.44.243.130)

; outgoing trunk access
; NANP
; toll fraud blockers
;;exten => _1809NXXXXXX,1,Goto(unassigned,s,1);
;;exten => _809NXXXXXX,2,Goto(unassigned,s,1);
; general case
;;exten => _NXXNXXXXXX,2,Macro(diallink2voipNL,1${EXTEN})
;;exten => _1NXXNXXXXXX,1,Macro(diallink2voipNL,${EXTEN})
; international
;;exten => _011.,1,Macro(diallink2voipNL,${EXTEN})

exten => 2100,1,Macro(dialGSM,wiredPhone)
exten => 2101,1,Macro(dialGSM,softPhone)
exten => 2102,1,Macro(dialGSM,IMSI602022093727869)
exten => 2103,1,Macro(dialGSM,IMSI602030060770718)

;[sip-local]
;exten => 2100,1,Macro(dialGSM,wiredPhone)
;exten => 2101,1,Macro(dialGSM,softPhone)
;exten => 2102,1,Macro(dialGSM,IMSI602022093727869)
;exten => 2103,1,Macro(dialGSM,IMSI602030060770718)
[/code]

and I added to sip.conf

 [general]
; Comment these out if no backhaul is available.
; Use the pair with the shortest latency.
register => kestrel0:v01ptest@sip.ca1.link2voip.com:5060
register => kestrel0:v01ptest@sip.ca2.link2voip.com:5060
;register => kestrel0:v01ptest@sip.us1.link2voip.com:5060
;register => kestrel0:v01ptest@sip.us2.link2voip.com:5060
;register => kestrel0:v01ptest@sip.nl1.link2voip.com:5060
;register => kestrel0:v01ptest@sip.nl2.link2voip.com:5060
rtpstart=16386
rtpend=16482
relaxdtmf=yes

[wiredPhone]
callerid=5552102
canreinvite=no
type=friend
context=sip-external
allow=ulaw
allow=gsm
host=dynamic
dtmfmode=auto




[IMSI602022093727869] ; samsung gt-xxxx
callerid=2111
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info


[IMSI602030060770718] ; H's Motorolla
callerid=2112
canreinvite=no
type=friend
context=sip-external
allow=gsm
host=dynamic
dtmfmode=info


[softPhone]
callerid=2101
canreinvite=no
type=friend
context=sip-external
allow=ulaw
allow=gsm
host=dynamic
;dtmfmode=info

you may laugh at my mistakes :smiley: , but this’s my 1st time here … so please try to help me
Thanks in advance

No reply to register would normally be a firewall or NAT problem.

I don’t know where OpenBTS fits in here, but as one user gave a strong impression they weren’t licensed, I would caution that you will normally need a government issued radio transmitting licence.