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<DIV><FONT face=Arial size=2>Hi everybody,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I am testing Optipoint 400 Standard SIP (Firmware
2.3.14) with Asterisk.</FONT></DIV>
<DIV><FONT face=Arial size=2>It is posible to dial from another Phone (x-lite)
to the Optipoint, but when I try to dial from the Optipoint there is no dialtone
and there is only a short beep when I dial Numbers.</FONT></DIV>
<DIV><FONT face=Arial size=2>The Optipoint shows "no Server..." (Registrar?) in
Display.</FONT></DIV>
<DIV><FONT face=Arial size=2>Sip debug shows no unusual (to me)
Messages.</FONT></DIV>
<DIV><FONT face=Arial size=2>Sip show peers:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> <FONT face=Arial size=2>Name/username
Host Dyn Nat
ACL Mask
Port
Status<BR>1006/1006
(Unspecified)
D 255.255.255.255
0
Unmonitored<BR>1005/1005
(Unspecified)
D 255.255.255.255
0
Unmonitored<BR>1004/1004
192.168.1.98
D 255.255.255.255
5060 Unmonitored ---This is the Optipoint
400<BR>sipgate/wendys
217.10.79.9
255.255.255.255 5060 Unmonitored<BR></FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Optipoint Config:</FONT></DIV>
<DIV><FONT face=Arial size=2>Registrar: 192.168.1.99</FONT></DIV>
<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2>SIP-Server: 192.168.1.99</FONT></DIV>
<DIV><FONT face=Arial size=2>Realm: 192.168.1.99</FONT></DIV>
<DIV>Routing = Server</DIV>
<DIV>register by Name (Tested also register by ID doesn't matter since they are
the same)</DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT></DIV></FONT>
<DIV><FONT face=Arial size=2>SIP conf:</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial
size=2>[1004]<BR>type=friend<BR>username=1004<BR>host=dynamic<BR>dtmfmode=rfc2833<BR>callerid="1004"
<1004><BR>mailbox=1000<BR>context=sip</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sip debug peer 1004:</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>SIP Debugging Enabled for IP:
192.168.1.98:5060<BR>Sending to 192.168.1.98 : 5060 (non-NAT)<BR>Transmitting
(no NAT):<BR>SIP/2.0 100 Trying<BR>Via: SIP/2.0/UDP
192.168.1.98:5060;branch=z9hG4bKa956fdf98<BR>From: 1004
<sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5<BR>To: 1004
<sip:1004@192.168.1.99>;tag=as50ba5e89<BR>Call-ID: <A
href="mailto:8003812aded555fef6f5827f4a12298b@192.168.1.99">8003812aded555fef6f5827f4a12298b@192.168.1.99</A><BR>CSeq:
847678061 REGISTER<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Contact: <sip:1004@192.168.1.99><BR>Content-Length:
0<BR>
<BR>
<BR> to 192.168.1.98:5060<BR>Transmitting (no NAT):<BR>SIP/2.0 200
OK<BR>Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98<BR>From: 1004
<sip:1004@192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5<BR>To: 1004
<sip:1004@192.168.1.99>;tag=as50ba5e89<BR>Call-ID: <A
href="mailto:8003812aded555fef6f5827f4a12298b@192.168.1.99">8003812aded555fef6f5827f4a12298b@192.168.1.99</A><BR>CSeq:
847678061 REGISTER<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER<BR>Expires: 3600<BR>Contact:
<sip:1004@192.168.1.99>;expires=3600<BR>Date: Sun, 27 Jun 2004 18:26:39
GMT<BR>Content-Length:
0<BR>
<BR>
<BR> to 192.168.1.98:5060<BR>Scheduling destruction of call <A
href="mailto:'8003812aded555fef6f5827f4a12298b@192.168.1.99'">'8003812aded555fef6f5827f4a12298b@192.168.1.99'</A>
in 15000 ms<BR>11 headers, 2 lines<BR>Reliably Transmitting:<BR>NOTIFY
sip:1004@192.168.1.98 SIP/2.0<BR>Via: SIP/2.0/UDP
192.168.1.99:5060;branch=z9hG4bK57fa7f3b;rport<BR>From: "asterisk"
<sip:asterisk@192.168.1.99>;tag=as5071967c<BR>To:
<sip:1004@192.168.1.98><BR>Contact:
<sip:asterisk@192.168.1.99><BR>Call-ID: <A
href="mailto:2d9992ad78c69f110a8557a74745c333@192.168.1.99">2d9992ad78c69f110a8557a74745c333@192.168.1.99</A><BR>CSeq:
102 NOTIFY<BR>User-Agent: Asterisk PBX<BR>Event:
message-summary<BR>Content-Type:
application/simple-message-summary<BR>Content-Length:
36<BR>
<BR>Messages-Waiting: no<BR>Voicemail: 0/0<BR> (no NAT) to
192.168.1.98:5060<BR>Scheduling destruction of call <A
href="mailto:'2d9992ad78c69f110a8557a74745c333@192.168.1.99'">'2d9992ad78c69f110a8557a74745c333@192.168.1.99'</A>
in 15000 ms<BR>Destroying call <A
href="mailto:'2d9992ad78c69f110a8557a74745c333@192.168.1.99'">'2d9992ad78c69f110a8557a74745c333@192.168.1.99'</A><BR>Destroying
call <A
href="mailto:'8003812aded555fef6f5827f4a12298b@192.168.1.99'">'8003812aded555fef6f5827f4a12298b@192.168.1.99'</A></FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>There is no event on hookoff, but there is still no
event at the Softphone that workes fine!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Could anybody help?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>With best regards</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Marco Wendenburg</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2> </DIV></FONT></BODY></HTML>