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