Re: [reSIProcate-users] Sending INVITE as Server
Apparently, looking at the log would have helped (Although at first
glance it doesn't seem obvious why no Via is detected in the message -
I'll keep digging...):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bK-d8754z-1d74ed4a0b72065e-1---d8754z-
Contact: <sip:201@xxxxxxxxxxxxx:5060>
To: <sip:201@xxxxxxxxxxxxx>;tag=8f7f65c1-698965
From: <sip:111@xxxxxxxxxxx:5060>;tag=e37ac45c
Call-ID: D1B9-11C0-466989655479F0AC8BD1-009@SipHost
CSeq: 17 INVITE
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, INFO, PRACK, REFER, SUBSCRIBE,
NOTIFY, UPDATE
Content-Type: application/sdp
User-Agent: dlink 12-38-25912059-0.10.21-DSLX
Content-Length: 363
v=0
o=201 1806175820 1806175820 IN IP4 192.168.0.175
s=Session SDP
c=IN IP4 192.168.0.175
t=0 0
m=image 9000 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
DEBUG | 20090105-151609.390 | Resiprogate | RESIP:DUM | 4168 |
Dialog.cxx:328 | Dialog::dispatch: SipResp: 200 tid=1d74ed4a0b72065e
cseq=INVITE contact=201@xxxxxxxxxxxxx:5060 / 17 from(wire)
DEBUG | 20090105-151609.390 | Resiprogate | RESIP | 4168 |
SipMessage.cxx:972 | SipMessage::getContents: application/sdp
DEBUG | 20090105-151609.390 | Resiprogate | RESIP | 4168 |
Helper.cxx:2119 | Got sdp
INFO | 20090105-151609.390 | Resiprogate | RESIP | 4168 |
SipMessage.cxx:313 | Bad message with no Vias: Content-Length: 0
DEBUG | 20090105-151609.390 | Resiprogate | RESIP | 4168 |
baseexception.cxx:17 | BaseException at .\SipMessage.cxx:314 No Via in
message
ERR | 20090105-151609.421 | Resiprogate | RESIP:DUM | 4168 |
DialogUsageManager.cxx:1367 | Illegal message rejected: No Via in message
Amnon David wrote:
Hi,
I'm sure what I'm trying to do is quite standard, but I can't seem to
find any reference to how to achieve it in resiprocate:
The basic idea is receiving a fax - in a nutshell, the gateway
application receives an INVITE from the remote endpoint, the codec is
negotiated and then starts transmitting audio to the telephony
endpoint. Shortly afterwards, the gateway application realizes that
the telephone is really a fax machine and therefore initiates an
INVITE with an SDP of T.38 back to the VoIP endpoint that initiated
the call. At this point the VoIP endpoint, sends back an OK - but the
problem is that this OK in not propagated to the resip application's
InviteSessionHandler callbacks, so there is no way for my application
to ACK it.
I can send resip stack and wireshark logs (where everything works fine
until the OK response for the fax INVITE), but just wanted to make
sure that I didn't miss a trivial solution before wasting bandwidth...
Thanks for any help,
Amnon