----- Original Message -----
Sent: Sunday, June 27, 2004 8:43 PM
Subject: [Asterisk-Users] Optipoint 400
Standard Sip
Hi everybody,
I am testing Optipoint 400 Standard SIP (Firmware
2.3.14) with Asterisk.
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.
The Optipoint shows "no Server..." (Registrar?)
in Display.
Sip debug shows no unusual (to me)
Messages.
Sip show peers:
Name/username
Host Dyn Nat
ACL
Mask
Port
Status
1006/1006
(Unspecified)
D 255.255.255.255
0
Unmonitored
1005/1005
(Unspecified)
D 255.255.255.255
0
Unmonitored
1004/1004
192.168.1.98
D 255.255.255.255
5060 Unmonitored ---This is the Optipoint
400
sipgate/wendys
217.10.79.9
255.255.255.255 5060
Unmonitored
Optipoint Config:
Registrar: 192.168.1.99
SIP-Server: 192.168.1.99
Realm: 192.168.1.99
Routing = Server
register by Name (Tested also register by ID doesn't matter since they
are the same)
SIP conf:
[1004]
type=friend
username=1004
host=dynamic
dtmfmode=rfc2833
callerid="1004"
<1004>
mailbox=1000
context=sip
Sip debug peer 1004:
SIP Debugging Enabled for IP:
192.168.1.98:5060
Sending to 192.168.1.98 : 5060 (non-NAT)
Transmitting
(no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.98:5060;branch=z9hG4bKa956fdf98
From: 1004
<sip:[EMAIL PROTECTED]>;tag=e4a7266f4d69369;epid=SC2026F5
To: 1004
<sip:[EMAIL PROTECTED]>;tag=as50ba5e89
Call-ID: [EMAIL PROTECTED]
CSeq:
847678061 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER
Contact:
<sip:[EMAIL PROTECTED]>
Content-Length:
0
to 192.168.1.98:5060
Transmitting (no NAT):
SIP/2.0 200
OK
Via: SIP/2.0/UDP 192.168.1.98:5060;branch=z9hG4bKa956fdf98
From: 1004
<sip:[EMAIL PROTECTED]>;tag=e4a7266f4d69369;epid=SC2026F5
To: 1004
<sip:[EMAIL PROTECTED]>;tag=as50ba5e89
Call-ID: [EMAIL PROTECTED]
CSeq:
847678061 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER
Expires: 3600
Contact:
<sip:[EMAIL PROTECTED]>;expires=3600
Date: Sun, 27 Jun 2004 18:26:39
GMT
Content-Length:
0
to 192.168.1.98:5060
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY
sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP
192.168.1.99:5060;branch=z9hG4bK57fa7f3b;rport
From: "asterisk"
<sip:[EMAIL PROTECTED]>;tag=as5071967c
To:
<sip:[EMAIL PROTECTED]>
Contact:
<sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq:
102 NOTIFY
User-Agent: Asterisk PBX
Event:
message-summary
Content-Type:
application/simple-message-summary
Content-Length:
36
Messages-Waiting: no
Voicemail: 0/0
(no NAT) to
192.168.1.98:5060
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms
Destroying call '[EMAIL PROTECTED]'
Destroying
call '[EMAIL PROTECTED]'
There is no event on hookoff, but there is still
no event at the Softphone that workes fine!
Could anybody help?
With best regards
Marco Wendenburg