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<div dir="ltr"><font color="#000000" size="2" face="Tahoma">Hi,</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="Tahoma"> <font face="tahoma">I used the downlaoded TAR ball and my calls worked, however, when upgrading to the SVN release...my SBC is rejecting the 200 OK (when the FS answers the call - using Conferencing app)..</font></font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr"><font size="2" face="tahoma">Here are teh bad and good 200 OK,,,,I see a lot of additional headers startin gwith X:FS , can I remove them?</font></div>
<div dir="ltr"><font size="2" face="tahoma"></font> </div>
<div dir="ltr">Bad one<br>
-----------------<br>
SIP/2.0 200 OK<br>
v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bK1662379c08490b63b0b9be6978d93ebe<br>
f: "SIMPLE SIGNAL "<sip:2222220208@x.x.x.20:5080;user=phone>;tag=945d1734da594444aac9d071edc798a7<br>
To: "trunkgroup user" <sip:1111114847@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=gX1Z34952QBpS<br>
i: <a href="mailto:674babc2dbf2a90da97bd68642865eae-23f978@x.x.x.210">674babc2dbf2a90da97bd68642865eae-23f978@x.x.x.210</a><br>
CSeq: 3240 INVITE<br>
Contact: <sip:1111114847@x.x.x.20:5080;transport=udp><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15225<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, NOTIFY<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 249<br>
X-FS-Display-Name: +11111114847<br>
X-FS-Display-Number: +11111114847<br>
X-FS-Support: update_display<br>
P-Asserted-Identity: "+11111114847" <+11111114847></div>
<div dir="ltr"> v=0<br>
o=FreeSWITCH 1256498762 1256498763 IN IP4 x.x.x.20<br>
s=FreeSWITCH<br>
c=IN IP4 x.x.x.20<br>
t=0 0<br>
m=audio 29406 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20</div>
<div dir="ltr">------------------------------</div>
<div dir="ltr">Good One<br>
-------------------------------<br>
SIP/2.0 200 OK<br>
v: SIP/2.0/UDP x.x.x.210:5060;branch=z9hG4bKffebde0efa06a6aa7772bae5ca085627<br>
f: "SIMPLE SIGNAL "<sip:2222220208@x.x.x.20:5080;user=phone>;tag=895fd35d4979e6e5b0cdcc795b8c8df2<br>
To: "trunkgroup user" <sip:1111114847@simplesignal.net;ssig=33402ab16f12e359b1f871b80bf7ff1a>;tag=KBXKrBQU99c6D<br>
i: <a href="mailto:28f0d47f1163e0d478247e6729762d98-23fedb@x.x.x.210">28f0d47f1163e0d478247e6729762d98-23fedb@x.x.x.210</a><br>
CSeq: 3278 INVITE<br>
Contact: <sip:1111114847@x.x.x.20:5080;transport=udp><br>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported<br>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO<br>
Supported: timer, precondition, path, replaces<br>
Allow-Events: talk, refer<br>
Content-Type: application/sdp<br>
Content-Disposition: session<br>
Content-Length: 249</div>
<div dir="ltr"> v=0<br>
o=FreeSWITCH 1256512215 1256512216 IN IP4 x.x.x.20<br>
s=FreeSWITCH<br>
c=IN IP4 x.x.x.20<br>
t=0 0<br>
m=audio 17376 RTP/AVP 0 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>
a=silenceSupp:off - - - -<br>
a=ptime:20</div>
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