Merged revisions 69794 via svnmerge from
[asterisk/asterisk.git] / channels / chan_sip.c
index 9d4472c..6b4834d 100644 (file)
@@ -15850,6 +15850,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
        if (dialog->owner->_state != AST_STATE_UP || dialog->redirip.sin_addr.s_addr)
                return;
 
+       /* If the call is involved in a T38 fax session do not check RTP timeout */
+       if (dialog->t38.state == T38_ENABLED)
+               return;
+
        /* If we have no timers set, return now */
        if (ast_rtp_get_rtpkeepalive(dialog->rtp) == 0 || (ast_rtp_get_rtptimeout(dialog->rtp) == 0 && ast_rtp_get_rtpholdtimeout(dialog->rtp) == 0))
                return;