<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML><HEAD><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=utf-8"></HEAD><BODY><DIV>hi, </DIV>
<DIV> my ATA186 confige as SIP(600) on my
Asterisk ,it only can be called in , but can not call
out .</DIV>
<DIV> between ATA186 and astersik there is
a VPN on two netscreen 5gt. </DIV>
<DIV> who can show me some idea ?</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> ATA 186 configure same as SIP.conf </DIV>
<DIV> </DIV>
<DIV> SIP.conf on Asterisk :</DIV>
<DIV> [general]<BR>port =
5060
; Port to bind to<BR>bindaddr =
0.0.0.0
; Address to bind to<BR>disallow=all<BR>allow=ulaw<BR>context =
local ; Default for local
calls</DIV>
<DIV>[600]<BR>type=friend<BR>username=600<BR>secret=monday<BR>host=dynamic<BR>defaultip=192.168.33.100<BR>canreinvite=no
; Cisco poops on reinvite
sometimes<BR>qualify=600
; Qualify peer is no more than 200ms away<BR>dtmfmode=rfc2833<BR>callerid = SZ
<600><BR>callgroup = 10<BR>pickupgroup = 10<BR>mailbox=600</DIV>
<DIV>[601]<BR>type=friend<BR>username=601<BR>secret=monday<BR>host=dynamic<BR>defaultip=192.168.33.100<BR>canreinvite=no
; Cisco poops on reinvite
sometimes<BR>qualify=600
; Qualify peer is no more than 200ms away<BR>dtmfmode=rfc2833<BR>callerid = SZ
<601><BR>callgroup = 10<BR>pickupgroup = 10<BR>mailbox=601</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>ON SIP debug mode shows:</DIV>
<DIV> <BR>to 192.168.33.100:5060<BR>Sip read: <BR>INVITE
sip:800@192.168.1.50;user=phone SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone><BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>Contact:
<sip:600@192.168.33.100:5060;user=phone;transport=udp><BR>User-Agent:
Cisco ATA 186 v3.1.0 atasip (040211A)<BR>Expires: 300<BR>Allow: ACK, BYE,
CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER<BR>Content-Length:
274<BR>Content-Type: application/sdp</DIV>
<DIV>v=0<BR>o=600 50100 50100 IN IP4 192.168.33.100<BR>s=ATA186 Call<BR>c=IN IP4
192.168.33.100<BR>t=0 0<BR>m=audio 10000 RTP/AVP 0 18 8 101<BR>a=rtpmap:0
PCMU/8000/1<BR>a=rtpmap:18 G729/8000/1<BR>a=fmtp:18 annexb=yes<BR>a=rtpmap:8
PCMA/8000/1<BR>a=rtpmap:101 telephone-event/8000<BR>a=fmtp:101 0-15</DIV>
<DIV>12 headers, 12 lines<BR>Ignoring this request<BR>Reliably Transmitting (no
NAT):<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="1220cba1"<BR>Content-Length: 0</DIV>
<DIV><BR> to 192.168.33.100:5060<BR>Retransmitting #1 (no NAT):<BR>SIP/2.0
407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="7e0a728d"<BR>Content-Length: 0</DIV>
<DIV>Leng9<BR> to 192.168.33.100:5060<BR>Retransmitting #2 (no
NAT):<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="0d6babf9"<BR>Content-Length: 0</DIV>
<DIV><BR>to 192.168.33.100:5060<BR>Retransmitting #1 (no NAT):<BR>SIP/2.0 407
Proxy Authentication Required<BR>Via: SIP/2.0/UDP 192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="1220cba1"<BR>Content-Length: 0</DIV>
<DIV>IP/2<BR> to 192.168.33.100:5060<BR>Retransmitting #2 (no
NAT):<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="7e0a728d"<BR>Content-Length: 0</DIV>
<DIV>Leng9<BR> to 192.168.33.100:5060<BR>Retransmitting #3 (no
NAT):<BR>SIP/2.0 407 Proxy Authentication Required<BR>Via: SIP/2.0/UDP
192.168.33.100:5060<BR>From:
<sip:600@192.168.1.50;user=phone>;tag=2459813530<BR>To:
<sip:800@192.168.1.50;user=phone>;tag=as74d2a1cb<BR>Call-ID: <A
href="mailto:3456456998@192.168.33.100">3456456998@192.168.33.100</A><BR>CSeq: 1
INVITE<BR>User-Agent: Asterisk PBX<BR>Contact: <BR>Proxy-Authenticate: Digest
realm="asterisk", nonce="0d6babf9"<BR>Content-Length: 0</DIV>
<DIV><BR>to 192.168.33.100:5060<BR>Sip read: <BR>REGISTER sip:192.168.1.50
SIP/2.0<BR>Via: SIP/2.0/UDP 192.168.1.58:5060<BR>From:
<sip:827@192.168.1.50;user=phone>;tag=1707128448<BR>To:
<sip:827@192.168.1.50;user=phone><BR>Call-ID: <A
href="mailto:386571196@192.168.1.58">386571196@192.168.1.58</A><BR>CSeq: 273
REGISTER<BR>Contact:
<sip:827@192.168.1.58:5060;user=phone;transport=udp>;expires=120<BR>User-Agent:
Cisco ATA 186 v3.1.0 atasip (040211A)<BR>Content-Length: 0</DIV>
<DIV> </DIV></BODY></HTML>