Hello,

My locale UAs do not know anything about T.38 protocol and I want to send a
fax to some PSTN subscriber via OVH, my french VoIP provider, in an RTP/PCMA
session. This session begins normally but, as soon as OVH's gateway
recognizes fax tones from remote and/or locale side, an undesirable UPDATE
with SDP for a T.38 session is sent to my side. The problem is that,
immediately, FreeSwitch says "200 OK" instead of something like "488 Not
acceptable here" and this is the beginning of a dialogue of the deaf which
only takes end when remote fax-machine hangs up.

I have done several things that I found in the WIKI:

In my dialplan added these lines:

      <action application="set" data="sip_ignore_reinvites=true"/>
      <action application="set" data="fax_enable_t38=false"/>
      <action application="set" data="fax_enable_t38_request=false"/>

In ~/conf/vars.xml:

     <X-PRE-PROCESS cmd="set" data="refuse_t38=true"/>

In ~/conf/sip_profiles/external/sip.ovh.fr.xml:

     <param name="t38-passthru" value="false"/>

But without better result.


This is an example of the UPDATE request:

UPDATE sip:gw+sip.ovh.fr@92.146.52.82:5080;transport=udp;gw=sip.ovh.fr SIP/2.0
Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4
Contact: <sip:10.7.1.60:5060>
Content-Type: application/sdp
Cseq: 29265730 UPDATE
From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300
Max-Forwards: 30
Record-Route: <sip:91.121.129.20:5060;lr>
To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77
User-Agent: Cirpack/v4.56 (gw_sip)
Content-Length: 299

v=0
o=cp10 138814832168 138814832170 IN IP4 10.7.1.132
s=SIP Call
c=IN IP4 91.121.128.144
t=0 0
m=image 35996 udptl t38
a=sendrecv
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:TransferredTCF
a=T38FaxMaxBuffer:1000
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy

And that is the OK answer:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.129.20:5060;branch=z9hG4bK-DIVG-481b17bf-03a84e77
Record-Route: <sip:91.121.129.20:5060;lr>
From: <sip:0556ZZZZZZ@sip.ovh.fr>;tag=00-08184-01c44a83-0e1305300
To: \"0033972XXXXXX\" <sip:0033972XXXXXX@sip.ovh.fr>;tag=75761tHD7541g
Call-ID: 9686bdc8-e997-1231-27aa-001fc61584f4
Cseq: 29265730 UPDATE
Contact: <sip:gw+sip.ovh.fr@92.146.52.82:5080>;transport=udp;gw=sip.ovh.fr
User-Agent: FreeSWITCH-mod_sofia/1.5.8b+git~20131213T181356Z~87751f9eaf~64bit
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY
Supported: timer, precondition, path, replaces
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 201

v=0
o=FreeSWITCH 1388128642 1388128643 IN IP4 92.146.52.82
s=FreeSWITCH CH
c=IN IP4 92.146.52.82
t=0 0
m=audio 19678 RTP/AVP 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

Any help much appreciated, thanks in advance...







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