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hi:<br>I am using GSM gateway to regiester to freeswitch, the incoming from gsm gateway is ok, but i can not regiester<br>into freeswitch due to the gateway connection.<br>===the debug info from fs_CLI<br> ------------------------------------------------------------------------<br>2011-09-21 23:14:50.645350 [ERR] sofia_reg.c:1496 gsm Registration Failed with status Request Timeout [408]. failure #33<br>2011-09-21 23:14:53.318802 [WARNING] sofia_reg.c:386 gsm Failed Registration, setting retry to 1020 seconds.<br>=====sip trace ===============<br><br> ------------------------------------------------------------------------<br>send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:50.147741:<br> ------------------------------------------------------------------------<br> REGISTER sip:172.16.33.11 SIP/2.0<br> Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H<br> Max-Forwards: 70<br> From: <sip:1000@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr<br> To: <sip:1000@172.16.33.11;transport=udp><br> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6<br> CSeq: 17972965 REGISTER<br> Contact: <sip:gw+gsm@172.16.33.100:5080;transport=udp;gw=gsm><br> Expires: 30<br> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br> Supported: timer, precondition, path, replaces<br> Content-Length: 0<br> ------------------------------------------------------------------------<br>send 590 bytes to udp/[172.16.33.11]:5060 at 15:14:46.146854:<br> ------------------------------------------------------------------------<br> REGISTER sip:172.16.33.11 SIP/2.0<br> Via: SIP/2.0/UDP 172.16.33.100:5080;rport;branch=z9hG4bKB09cFa6Uvyc1H<br> Max-Forwards: 70<br> From: <sip:1000@172.16.33.11;transport=udp>;tag=B1ytyyXQS3NKr<br> To: <sip:1000@172.16.33.11;transport=udp><br> Call-ID: 4809dacc-aeee-4a33-a245-884c6788a4a6<br> CSeq: 17972965 REGISTER<br> Contact: <sip:gw+gsm@172.16.33.100:5080;transport=udp;gw=gsm><br> Expires: 30<br> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br> Supported: timer, precondition, path, replaces<br> Content-Length: 0<br><br>=======gateway confi: ======<br><include><br> <gateway name="gsm"><br> <!--/// account username *required* ///--><br> <!-- Extension 1000 in my case --><br> <param name="username" value="1000"/><br> <!--/// auth realm: *optional* same as gateway name, if blank ///--><br> <!-- It must aim to the GateWay IP 192.168.2.2 in my case --><br> <b> <param name="realm" value="172.16.33.11"/></b><br> <!--1234 in my case--><br> <param name="password" value="1234"/><br> <!--/// expire in seconds: *optional* 3600, if blank ///--><br> <param name="expire-seconds" value=""/><br> <!--/// do not register ///--><br> <param name="register" value="false"/><br> <!-- which transport to use for register --><br> <param name="sip-trace" value="yes"/><br><br> </gateway><br> </include><br>freeswitch@internal> sofia status<br> Name Type Data State<br>=================================================================================================<br> external profile sip:mod_sofia@172.16.33.100:5080 RUNNING (0)<br> external::example.com gateway sip:joeuser@example.com NOREG<br> <b>external::gsm gateway sip:1000@172.16.33.11 FAIL_WAIT</b><br> internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0)<br> internal profile sip:mod_sofia@172.16.33.100:5060 RUNNING (0)<br> 172.16.33.100 alias internal ALIASED<br>=================================================================================================<br>somehow, the gateway never sends anything. anyone can give a hit?<br><br>Best regards,<br>James.zhu<br>Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri<->SIP).<br>website: www.voipviews.com <br> </div></body>
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