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[lei.tlfly at gmail.com]

Hi all, I get a  "415 Unsupported Media Type" when FS receiving call from a softswitch. I captured some packets, It seems that the softswitch use SIP-I protocol, does FS can handle SIP-I message?
    ===here is the invite messagefrom softswitch
INVITE sip:xxxxx@xxxx:5060;user=phone SIP/2.0
Contact: <sip:xxxx@xxxx:5061>
MIME-version: 1.0
Content-Type: multipart/mixed;boundary=Alcatel-boundary
To: <sip:xxxx@xxxx:5060;user=phone>
From: xxxx<sip:xxxx@xxxxx:5061;user=phone>;tag=73D332463135364195291201
P-Asserted-Identity: <sip:xxxx@xxxxx:5061;user=phone>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE
Supported: 100rel,timer,replaces,diversion
Expires: 155
Session-Expires: 1800
Min-SE: 90
Call-ID: 01FD034872814000000230A1@sip-3
Max-Forwards: 70
CSeq: 1 INVITE
Timestamp: 10645
Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
Content-Length: 542

--Alcatel-boundary
Content-Type: application/sdp

v=0
o=- 5 8 IN IP4 xxxxxx
s=SDP Data
c=IN IP4 xxxxx
t=0 0
m=audio 10266 RTP/AVP 0 8 96 110 111 112 113 3 97
a=rtpmap:110 speex/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 TELEPHONE-EVENT/8000
a=fmtp:110 mode=3
a=rtpmap:111 speex/8000/1
a=rtpmap:3 GSM/8000
a=ptime:30
a=fmtp:111 mode=2
a=rtpmap:112 speex/8000/1

--Alcatel-boundary
Content-Type: application/ISUP;version=N/A

.. ....
...0.D...
...V..0......1...n..p...
--Alcatel-boundary--

=====response message from FS
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP xxxxx:5061;branch=z9hG4bK8E1558EA4F1BE09A2BCB669331A9AC7E
From: xxxxx <sip:xxxxx@xxxx:5061;user=phone>;tag=73D332463135364195291201
To: <sip:xxxxx@xxxxx:5060;user=phone>;tag=m6XS44BKF4pSS
Call-ID: 01FD034872814000000230A1@sip-3
CSeq: 1 INVITE
Timestamp: 10645 0.000000
User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460
Accept: application/sdp
Accept-Encoding:
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Allow-Events: talk, refer
Content-Length: 0
--
Lei.Tang
lei.tlfly@gmail.com (lei.tlfly@gmail.com)

[lei.tlfly at gmail.com]

FYI, Here is the log when I set sofia loglevel all 9
==============
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 01B2E0C0 from (udp/MyIP:5060) has 1315 bytes, veclen
= 1
tport(00DFE3E8): msg 01B2E0C0 (1315 bytes) from udp/SSIP:5060/sip next=000
00000
nta: received INVITE sip:AAA@MyIP:5060;user=phone SIP/2.0 (CSeq
1)
nta: canonizing sip:AAA@MyIP:5060 with contact
nta: INVITE (1) going to a default leg
nta: timer shortened to 200 ms
tport_tsend(00DFE3E8) tpn = UDP/SSIP:5061
tport_resolve addrinfo = SSIP:5061
tport(00DFE3E8): not found by name UDP/SSIP:5061
tport_vsend(00DFE3E8): 652 bytes of 652 to udp/SSIP:5061
tport_vsend returned 652
nta: sent 415 Unsupported Media Type for INVITE (1)
tport_wakeup_pri(00DFE3E8): events IN
tport_recv_event(00DFE3E8)
tport(00DFE3E8) msg 0192C510 from (udp/MyIP:5060) has 388 bytes, veclen =
 1
tport(00DFE3E8): msg 0192C510 (388 bytes) from udp/SSIP:5060/sip next=0000
0000
nta: received ACK sip:AAA@MyIP:5060;user=phone SIP/2.0 (CSeq 1)
nta: ACK (1) is going to INVITE (1)
nta: timer set next to 937 ms
nta: timer J fired, terminate 200 response
incoming_reclaim_all(00000000, 00000000, 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 3859 ms
nta: timer I fired, terminate 415 response
incoming_reclaim_all(00000000, 00000000, 02E2FEB8)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set

2009/11/2 Lei Tang <lei.tlfly@gmail.com (lei.tlfly@gmail.com)>