[Asterisk-Users] ATA186 reguest problem

[Weiming Jiang]

hi, 
   my   ATA186 confige as  SIP(600)  on my Asterisk ,it  only can be called  in , but  can not  call out .
   between ATA186 and astersik  there is  a  VPN on two netscreen 5gt.   
    who can show me  some idea ?
 
 
 ATA 186  configure same as SIP.conf 
 
  SIP.conf  on  Asterisk :
  [general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
disallow=all
allow=ulaw
context = local         ; Default for local calls
[600]
type=friend
username=600
secret=monday
host=dynamic
defaultip=192.168.33.100
canreinvite=no                  ; Cisco poops on reinvite sometimes
qualify=600                     ; Qualify peer is no more than 200ms away
dtmfmode=rfc2833
callerid = SZ <600>
callgroup = 10
pickupgroup = 10
mailbox=600
[601]
type=friend
username=601
secret=monday
host=dynamic
defaultip=192.168.33.100
canreinvite=no                  ; Cisco poops on reinvite sometimes
qualify=600                     ; Qualify peer is no more than 200ms away
dtmfmode=rfc2833
callerid = SZ <601>
callgroup = 10
pickupgroup = 10
mailbox=601
 
 
 
 
 
ON  SIP debug mode shows:
   
to 192.168.33.100:5060
Sip read: 
INVITE sip:800 at 192.168.1.50;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
Contact: <sip:600 at 192.168.33.100:5060;user=phone;transport=udp>
User-Agent: Cisco ATA 186  v3.1.0 atasip (040211A)
Expires: 300
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER
Content-Length: 274
Content-Type: application/sdp
v=0
o=600 50100 50100 IN IP4 192.168.33.100
s=ATA186 Call
c=IN IP4 192.168.33.100
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 12 lines
Ignoring this request
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="1220cba1"
Content-Length: 0

 to 192.168.33.100:5060
Retransmitting #1 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="7e0a728d"
Content-Length: 0
Leng9
 to 192.168.33.100:5060
Retransmitting #2 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="0d6babf9"
Content-Length: 0

to 192.168.33.100:5060
Retransmitting #1 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="1220cba1"
Content-Length: 0
IP/2
 to 192.168.33.100:5060
Retransmitting #2 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="7e0a728d"
Content-Length: 0
Leng9
 to 192.168.33.100:5060
Retransmitting #3 (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600 at 192.168.1.50;user=phone>;tag=2459813530
To: <sip:800 at 192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998 at 192.168.33.100
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Proxy-Authenticate: Digest realm="asterisk", nonce="0d6babf9"
Content-Length: 0

to 192.168.33.100:5060
Sip read: 
REGISTER sip:192.168.1.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.58:5060
From: <sip:827 at 192.168.1.50;user=phone>;tag=1707128448
To: <sip:827 at 192.168.1.50;user=phone>
Call-ID: 386571196 at 192.168.1.58
CSeq: 273 REGISTER
Contact: <sip:827 at 192.168.1.58:5060;user=phone;transport=udp>;expires=120
User-Agent: Cisco ATA 186  v3.1.0 atasip (040211A)
Content-Length: 0
 
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