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@192.168.1.50;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.33.100:5060
From: <sip:600@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>
Call-ID: 3456456998@192.168.33.100
CSeq: 1 INVITE
Contact: <sip:600@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=2459813530
To: <sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb
Call-ID: 3456456998@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@192.168.1.50;user=phone>;tag=1707128448
To: <sip:827@192.168.1.50;user=phone>
Call-ID: 386571196@192.168.1.58
CSeq: 273 REGISTER
Contact: <sip:827@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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