[Freeswitch-users] please help for Failed Registration, setting retry to 1020 seconds

[James zhu]

hi:
I am using GSM gateway to regiester to freeswitch, the incoming from gsm gateway is ok, but i can not regiester
into freeswitch due to the gateway connection.
===the debug info from fs_CLI
   ------------------------------------------------------------------------
2011-09-21 23:14:50.645350 [ERR] sofia_reg.c:1496 gsm Registration Failed with status Request Timeout [408]. failure #33
2011-09-21 23:14:53.318802 [WARNING] sofia_reg.c:386 gsm Failed Registration, setting retry to 1020 seconds.
=====sip trace ===============

   ------------------------------------------------------------------------
send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:50.147741:
   ------------------------------------------------------------------------
   REGISTER sip:172.16.33.11 SIP/2.0
   Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
   Max-Forwards: 70
   From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
   To: <sip:1000 at 172.16.33.11;transport=udp>
   Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
   CSeq: 17972965 REGISTER
   Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
   Expires: 30
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   ------------------------------------------------------------------------
send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:46.146854:
   ------------------------------------------------------------------------
   REGISTER sip:172.16.33.11 SIP/2.0
   Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H
   Max-Forwards: 70
   From: <sip:1000 at 172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr
   To: <sip:1000 at 172.16.33.11;transport=udp>
   Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6
   CSeq: 17972965 REGISTER
   Contact: <sip:gw+gsm at 172.16.33.100:5080;transport=udp;gw=gsm>
   Expires: 30
   User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0

=======gateway confi: ======
<include>
  <gateway name="gsm">
  <!--/// account username *required* ///-->
  <!-- Extension 1000 in my case -->
  <param name="username" value="1000"/>
  <!--/// auth realm: *optional* same as gateway name, if blank ///-->
  <!-- It must aim to the  GateWay IP 192.168.2.2 in my case -->
  <param name="realm" value="172.16.33.11"/>
  <!--1234 in my case-->
  <param name="password" value="1234"/>
  <!--/// expire in seconds: *optional* 3600, if blank ///-->
  <param name="expire-seconds" value=""/>
  <!--/// do not register ///-->
  <param name="register" value="false"/>
  <!-- which transport to use for register -->
  <param name="sip-trace" value="yes"/>

 </gateway>
 </include>
freeswitch at internal> sofia status
                     Name          Type                                       Data      State
=================================================================================================
                 external       profile           sip:mod_sofia at 172.16.33.100:5080      RUNNING (0)
    external::example.com       gateway                    sip:joeuser at example.com      NOREG
            external::gsm       gateway                      sip:1000 at 172.16.33.11      FAIL_WAIT
            internal-ipv6       profile                   sip:mod_sofia@[::1]:5060      RUNNING (0)
                 internal       profile           sip:mod_sofia at 172.16.33.100:5060      RUNNING (0)
            172.16.33.100         alias                                   internal      ALIASED
=================================================================================================
somehow, the gateway never sends anything.  anyone can give a hit?

Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).
website: www.voipviews.com 
 		 	   		  
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