Asterisk question

[Travis Roy]

> what does your sip.conf look like?


; Note: If your SIP devices are behind a NAT and your Asterisk
;  server isn't, try adding "nat=1" to each peer definition to
;  solve translation problems.

[general]

port = 5060           ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf



> 
> Go into asterisk - asterisk -r
> 
> At the command prompt enter:
> set debug 4
> set verbose 4
> 
> Watch the REGISTER come in, cut and paste and send to me.


Here's the whole thing:

Sip read:
REGISTER sip:phone.scootz.net SIP/2.0
Via: SIP/2.0/UDP
63.251.138.229:33676;rport;branch=z9hG4bKCA8D950035794F579210A94EF67A7067
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>
Contact: "Travis Roy" <sip:200 at 63.251.138.229:33676>
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56481 REGISTER
Expires: 1800
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


11 headers, 0 lines
Using latest request as basis request
Sending to 63.251.138.229 : 33676 (non-NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
63.251.138.229:33676;branch=z9hG4bKCA8D950035794F579210A94EF67A7067;received=63.251.138.229;rport=33676
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56481 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200 at 69.84.130.62>
Content-Length: 0


 to 63.251.138.229:33676
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
63.251.138.229:33676;branch=z9hG4bKCA8D950035794F579210A94EF67A7067;received=63.251.138.229;rport=33676
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>;tag=as07b51428
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56481 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200 at 69.84.130.62>
WWW-Authenticate: Digest realm="asterisk", nonce="3dee6301"
Content-Length: 0


 to 63.251.138.229:33676
Scheduling destruction of call
'46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net' in 15000 ms
asterisk1*CLI>

Sip read:
REGISTER sip:phone.scootz.net SIP/2.0
Via: SIP/2.0/UDP
63.251.138.229:33676;rport;branch=z9hG4bKDBC56797DDE345BD849BEADCACE36149
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>
Contact: "Travis Roy" <sip:200 at 63.251.138.229:33676>
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56482 REGISTER
Expires: 1800
Authorization: Digest
username="200",realm="asterisk",nonce="3dee6301",response="54b3531a2562df6043bf91260d6dea06",uri="sip:phone.scootz.net"
Max-Forwards: 70
User-Agent: X-Lite release 1103m
Content-Length: 0


12 headers, 0 lines
Using latest request as basis request
Sending to 63.251.138.229 : 33676 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
63.251.138.229:33676;branch=z9hG4bKDBC56797DDE345BD849BEADCACE36149;received=63.251.138.229;rport=33676
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56482 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200 at 69.84.130.62>
Content-Length: 0


 to 63.251.138.229:33676
Transmitting (NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
63.251.138.229:33676;branch=z9hG4bKDBC56797DDE345BD849BEADCACE36149;received=63.251.138.229;rport=33676
From: Travis Roy <sip:200 at phone.scootz.net>;tag=217828604
To: Travis Roy <sip:200 at phone.scootz.net>;tag=as07b51428
Call-ID: 46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net
CSeq: 56482 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:200 at 69.84.130.62>
Content-Length: 0


 to 63.251.138.229:33676
Scheduling destruction of call
'46284E07B31F41E38B9DE08669B10B1E at phone.scootz.net' in 15000 ms













> 
> Ed
> 
> Travis Roy wrote:
> > So I found Asterisk at Home while doing some looking into VoIP stuff.
> > 
> > I set it up (very easy) but I'm having a problem.
> > 
> > I load up a softphone on my desktop and try to connect and I get this error:
> > 
> > Jul 14 12:41:10 NOTICE[1579]: Registration from 'Travis Roy
> > <sip:200 at phone.scootz.net>' failed for '69.84.130.55'
> > 
> > 200 is my extension.
> > _______________________________________________
> > gnhlug-discuss mailing list
> > gnhlug-discuss at mail.gnhlug.org
> > http://mail.gnhlug.org/mailman/listinfo/gnhlug-discuss
> > 
> _______________________________________________
> gnhlug-discuss mailing list
> gnhlug-discuss at mail.gnhlug.org
> http://mail.gnhlug.org/mailman/listinfo/gnhlug-discuss
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