[Asterisk-Users] Optipoint 400 Standard Sip

Fri Jul 2 10:03:25 MST 2004

Hi,

nobody got any Idea?

;-(
  ----- Original Message ----- 
  From: wendys 
  To: Asterisk-Users 
  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: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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