[Asterisk-Users] Optipoint 400 Standard Sip

[wendys]

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:1004 at 192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5
To: 1004 <sip:1004 at 192.168.1.99>;tag=as50ba5e89
Call-ID: 8003812aded555fef6f5827f4a12298b at 192.168.1.99
CSeq: 847678061 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1004 at 192.168.1.99>
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:1004 at 192.168.1.99>;tag=e4a7266f4d69369;epid=SC2026F5
To: 1004 <sip:1004 at 192.168.1.99>;tag=as50ba5e89
Call-ID: 8003812aded555fef6f5827f4a12298b at 192.168.1.99
CSeq: 847678061 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: <sip:1004 at 192.168.1.99>;expires=3600
Date: Sun, 27 Jun 2004 18:26:39 GMT
Content-Length: 0
                                                                                                                
                                                                                                                
 to 192.168.1.98:5060
Scheduling destruction of call '8003812aded555fef6f5827f4a12298b at 192.168.1.99' in 15000 ms
11 headers, 2 lines
Reliably Transmitting:
NOTIFY sip:1004 at 192.168.1.98 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.99:5060;branch=z9hG4bK57fa7f3b;rport
From: "asterisk" <sip:asterisk at 192.168.1.99>;tag=as5071967c
To: <sip:1004 at 192.168.1.98>
Contact: <sip:asterisk at 192.168.1.99>
Call-ID: 2d9992ad78c69f110a8557a74745c333 at 192.168.1.99
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 '2d9992ad78c69f110a8557a74745c333 at 192.168.1.99' in 15000 ms
Destroying call '2d9992ad78c69f110a8557a74745c333 at 192.168.1.99'
Destroying call '8003812aded555fef6f5827f4a12298b at 192.168.1.99'



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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