[Asterisk-Users] Optipoint 400 Standard Sip

Sun, 27 Jun 2004 11:49:50 -0700

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
 
 
 
 

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