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Hi,<BR>
<BR>
i have the following scenario when sending a fax:<BR>
ATA => FS => Provider => destination<BR>
<BR>
<BR>
The ATA tries to send a fax to destination and in the middle of the call there is the re-invite in t38 from the provider. <BR>
In dialplan i have: <action application="export" data="t38_passthru=true"/><BR>
<BR>
<BR>
The invite from the provider is:<BR>
<BR>
INVITE <A HREF="sip:gw+TRUNKIP@10.10.1.254:5060">sip:gw+TRUNKIP@10.10.1.254:5060</A>;transport=udp;gw=TRUNKIP SIP/2.0<BR>
Via: SIP/2.0/UDP 10.10.1.1:5060;branch=z9hG4bKno6j3r305os1nm4ph2a1sb0000010.1<BR>
Call-ID: badda7ee-3980-1231-8db4-b4b52f5e67f8<BR>
From: <<A HREF="sip:4774@10.10.1.1:5060">sip:4774@10.10.1.1:5060</A>>;tag=dg2aai02-CC-52<BR>
To: "5313" <<A HREF="sip:nousername@10.10.1.254:5060">sip:nousername@10.10.1.254:5060</A>>;tag=t6DB42KFXarmN<BR>
CSeq: 2 INVITE<BR>
Contact: <<A HREF="sip:4774@10.10.1.1:5060">sip:4774@10.10.1.1:5060</A>;transport=udp><BR>
Max-Forwards: 69<BR>
Content-Length: 437<BR>
Content-Type: application/sdp<BR>
<BR>
v=0<BR>
o=HuaweiSoftX3000 38317113 38317116 IN IP4 10.10.1.2<BR>
s=Sip Call<BR>
c=IN IP4 10.10.1.2<BR>
t=0 0<BR>
m=image 29090 udptl t38<BR>
a=T38FaxVersion:0<BR>
a=T38MaxBitRate:14400<BR>
a=T38FaxRateManagement:transferredTCF<BR>
a=T38FaxUdpEC:t38UDPRedundancy<BR>
m=audio 34264 RTP/AVP 8 0 101<BR>
a=rtpmap:8 PCMA/8000<BR>
a=rtpmap:0 PCMU/8000<BR>
a=rtpmap:101 telephone-event/8000<BR>
a=ptime:20<BR>
a=silenceSupp:off - - - -<BR>
a=ecan:fb on -<BR>
a=X-fax<BR>
a=fmtp:101 0-15<BR>
<BR>
<BR>
And the invite sent to the ATA is:<BR>
<BR>
INVITE <A HREF="sip:5313@192.168.1.97:5060">sip:5313@192.168.1.97:5060</A>;ref=5313 SIP/2.0<BR>
Via: SIP/2.0/UDP 10.10.1.254;rport;branch=z9hG4bK87p0gpcv4p31g<BR>
Max-Forwards: 70<BR>
From: <<A HREF="sip:4774@sipdomain.net">sip:4774@sipdomain.net</A>>;tag=mKN5ZFeQ19XDr<BR>
To: "5313" <<A HREF="sip:5313@sipdomain.net">sip:5313@sipdomain.net</A>>;tag=5de97158a7c051b0o0<BR>
Call-ID: <A HREF="mailto:6aa68878-1a412510@192.168.1.97">6aa68878-1a412510@192.168.1.97</A><BR>
CSeq: 44056889 INVITE<BR>
Contact: <<A HREF="sip:4774@10.10.1.254:5060">sip:4774@10.10.1.254:5060</A>;transport=udp><BR>
User-Agent: suvoz<BR>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<BR>
Supported: precondition, path, replaces<BR>
Content-Type: application/sdp<BR>
Content-Disposition: session<BR>
Content-Length: 291<BR>
<BR>
v=0<BR>
o=FreeSWITCH 1368772703 1368772705 IN IP4 10.10.1.254<BR>
s=FreeSWITCH<BR>
c=IN IP4 10.10.1.254<BR>
t=0 0<BR>
m=image 13196 udptl t38<BR>
a=T38FaxVersion:0<BR>
a=T38MaxBitRate:14400<BR>
a=T38FaxRateManagement:transferredTCF<BR>
a=T38FaxMaxBuffer:500<BR>
a=T38FaxMaxDatagram:500<BR>
a=T38FaxUdpEC:t38UDPRedundancy<BR>
<BR>
<BR>
As you can see the audio part is not send to the ata.<BR>
<BR>
I understand that is another "call" between the ATA and FS, but should FS send an invite to the ATA containing a audio part as well?<BR>
<BR>
I'm not sure that is a bug or Is the normal behavior? should i open a jira for it?<BR>
<BR>
<BR>
Searching for previous issues i found this, but there was no reply ...<BR>
<A HREF="http://freeswitch-users.2379917.n2.nabble.com/Doubts-about-T38-passthru-td7582305.html">http://freeswitch-users.2379917.n2.nabble.com/Doubts-about-T38-passthru-td7582305.html</A><BR>
<BR>
<BR>
Thanks for the help,<BR>
António<BR>
<BR>
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