please help for Failed Registration, setting retry to 1020 seconds
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
Hello James, are you using a Dinstar GSM Gateway or this is a another issue? ?? -I used to get the same error with my gateway but only when I forced registration so your profile doesn't make much sense to me. -Check your gateway configuration, check the port and IP where your FS is listening. (5080 is external by default in FS and you'll probably have to set it in your GW) -your <param name="expire-seconds" value=""/> is empty. Please let me know if you need some help, I have the feeling you've solved this by now. ;) Best Regards 2011/9/21 James zhu <zhulizhong@live.com>
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
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participants (2)
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James zhu
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jcgpoza gonzalez