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