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@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr To: <sip:1000@172.16.33.11;transport=udp> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6 CSeq: 17972965 REGISTER Contact: <sip:gw+gsm@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@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr To: <sip:1000@172.16.33.11;transport=udp> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6 CSeq: 17972965 REGISTER Contact: <sip:gw+gsm@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@internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia@172.16.33.100:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG external::gsm gateway sip:1000@172.16.33.11 FAIL_WAIT internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) internal profile sip:mod_sofia@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