<div dir="ltr">We can investigate that, Meanwhile you could use regex in your dialplan for this situation and set "sip_ignore_reinvites=true" to mitigate.<div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Fri, Aug 16, 2013 at 8:56 AM, François <span dir="ltr"><<a href="mailto:fdelawarde@wirelessmundi.com" target="_blank">fdelawarde@wirelessmundi.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello list,<br>
<br>
I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending<br>
both audio+image (T38) in the initial INVITE. In a simple bridge<br>
scenario, FS will just ignore the T38 part and setup an audio connection<br>
only with b-leg.<br>
<br>
My issue is later on if/when we receive a re-INVITE (ex: SIP timers<br>
refresh) with the same SDP (audio+T38), FS now considers it as FAX and<br>
sends a re-INVITE to b-leg with T38 only! In case the destination is<br>
voice only, the call hangs up.<br>
<br>
---<br>
<br>
Asterisk apparently had the same issue, they solved it by setting up<br>
both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL<br>
they just re-INVITE b-leg with T38.<br>
<br>
See: <a href="https://reviewboard.asterisk.org/r/208/" target="_blank">https://reviewboard.asterisk.org/r/208/</a><br>
Also: <a href="https://issues.asterisk.org/jira/browse/ASTERISK-11843" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-11843</a><br>
<br>
Is this the right way to go? (Also is it feasible in FreeSwitch)? I can<br>
add a JIRA if necessary.<br>
<br>
Thanks,<br>
François.<br>
<br>
<br>
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