<html><body><div style="color:#000; background-color:#fff; font-family:HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif;font-size:10pt"><div class="" style=""><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">Hello</span><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;"><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">We are experiencing some difficulties with T38 faxing.</span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><span style="color: rgb(51, 51,
51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">I have a Asterisk 11.5.0 with libss7 and Sangoma A104DE digital interface card . The operating system is Centos 6</span><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">We are using this server to terminate calls to Telco.</span><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><span style="color: rgb(51, 51, 51); font-family: 'Helvetica
Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">So calls are coming to asterisk from sip and we are sending calls to Telco with Dahdi. (It is a one way interconnection only from asterisk to telco ,not from telco to asterisk).</span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br class="" style=""></div><div class="" style="background-color: transparent;"><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><font color="#333333" face="Helvetica Neue, Helvetica, Arial, sans-serif" class="" style=""><span style="line-height: 18px;" class="">when t38 invite is received asterisk is sending SIP 200 ok with g711u,g711a,g729 codecs in SDP. </span></font></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family:
'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br class="" style=""></div><div class="" style="background-color: transparent;"><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><font color="#333333" face="Helvetica Neue, Helvetica, Arial, sans-serif" class="" style=""><span style="line-height: 18px;" class=""> i tried t38pt_udptl=no and setvar=FAXOPT(gateway)=no but still asterisk is not sending SIP 488 to t38 invite. instead asterisk is sending </span></font><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">SIP 200 ok with g711u,g711a,g729 codecs in SDP. </span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color:
transparent;"><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><br class="" style=""></span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class="">I got traces with TCPDUMP . Although asterisk is not sending SIP 200 with T38 in SDP, asterisk is sending and receiving T38 messages with remote side and receives t38 fax succesfully.</span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><span style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height:
18px;" class=""><br></span></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br style="color: rgb(51, 51, 51); font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; line-height: 18px;" class=""><br class="" style=""></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;"><br class="" style=""></div><div class="" style="color: rgb(51, 51, 51); font-size: 13px; font-family: 'Helvetica Neue', Helvetica, Arial, sans-serif; font-style: normal; background-color: transparent;">Any help will be
appreciated </div><div class="" style=""><br class="" style=""></div></div></body></html>