Sip.conf context:
[9254784865]
type=peer
host=dynamic
context=dialtone-broadband
username=9254784865
secret=9254784865
disallow=all
allow=ulaw
canreinvite=no

Trace: debug=4, verbose=255
*CLI> sip debug ip 192.168.0.58
<-- SIP read from 192.168.0.58:5060: 
REGISTER sip:pts-pbx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKacUrLgAmx
Max-Forwards: 70
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 5 REGISTER
Contact: <sip:9254784865@192.168.0.58;user=phone>;expires=180
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.200.371
Content-Length: 0


--- (13 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.58 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.58:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKacUrLgAmx;received=192.168.0.58
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9254784865@192.168.1.62>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.0.58:5060:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKacUrLgAmx;received=192.168.0.58
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>;tag=as52feef26
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 5 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9254784865@192.168.1.62>
WWW-Authenticate: Digest realm="asterisk", nonce="2ef2c18a"
Content-Length: 0


---
Scheduling destruction of call '2369331387qZuO@192.168.0.58' in 15000 ms

<-- SIP read from 192.168.0.58:5060: 
REGISTER sip:pts-pbx SIP/2.0
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKaczOrLgAU
Max-Forwards: 70
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 6 REGISTER
Authorization: Digest username="9254784865",realm="asterisk",nonce="2ef2c18a",uri="sip:pts-pbx",algorithm=MD5,response="7080a10ee4204cb68a3d821ead1df44d"
Contact: <sip:9254784865@192.168.0.58;user=phone>;expires=180
Supported: em,timer,replaces,path
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 180
User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.200.371
Content-Length: 0


--- (14 headers 0 lines)---
Using latest REGISTER request as basis request
Sending to 192.168.0.58 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.58:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKaczOrLgAU;received=192.168.0.58
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 6 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9254784865@192.168.1.62>
Content-Length: 0


---
Transmitting (no NAT) to 192.168.0.58:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.58;branch=z9hG4bKaczOrLgAU;received=192.168.0.58
From: <sip:9254784865@mp108-1>;tag=1c1224014541
To: <sip:9254784865@mp108-1>;tag=as52feef26
Call-ID: 2369331387qZuO@192.168.0.58
CSeq: 6 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Expires: 180
Contact: <sip:9254784865@192.168.0.58;user=phone>;expires=180
Date: Sun, 23 Oct 2005 00:32:41 GMT
Content-Length: 0


---
Scheduling destruction of call '2369331387qZuO@192.168.0.58' in 15000 ms
Beginning asterisk shutdown....
Executing last minute cleanups
  == Destroying musiconhold processes
Asterisk cleanly ending (2).