<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 class="" style="">Hello</span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="">Does anyone has a working libb7 server that able to handle T38 faxing.</span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal;
background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="">I am working on this for the last 10 days. But still no solution. </span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="">I am receiving the call from sip and sending to telco with
ss7. </span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span class="" style="">At the beginning of the call everything is fine ( i mean codec negotiation) . Then the remote sip side detects the fax tone that remote end sends and </span><span style="background-color: transparent;">Then the sip remote side sends the t38 invite and asterisk sends SIP 200 OK message with G711 and G729 codec in SDP.</span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial,
'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span style="background-color: transparent;"><br></span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span style="background-color: transparent;">My customer is saying that if i am sending SIP 200 OK to his T38 invite .in SDP of SIP 200 OK there should be only T38.</span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;" class=""><span style="background-color: transparent;"> </span></div><div style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal;
background-color: transparent;" class=""><span class="" style=""><br class="" style=""></span></div><div class="" style=""><span style="font-size: 13px;" class="">Also disabling t38 with t38pt_udptl=no didnt change anything. Still asterisk is sending </span><span style="font-size: 10pt;" class="">SIP 200 OK message with G711 and G729 codec in SDP as reply to T38 invite.</span></div><div class="" style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"><span style="font-size: 10pt;" class=""><br></span></div><div class="" style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"><span style="font-size: 10pt;" class=""><br></span></div><div class="" style="color: rgb(0, 0, 0);
font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"><span style="font-size: 10pt;" class="">I will be glad if current libss7 users can advice me a way to find a solution on this issue</span></div><div class="" style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"><span style="font-size: 10pt;" class=""><br></span></div><div class="" style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"><span style="font-size: 10pt;" class="">Regards</span></div><div class="" style="color: rgb(0, 0, 0); font-size: 10pt; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida
Grande', sans-serif; font-style: normal; background-color: transparent;"><br></div><div class="" style="color: rgb(0, 0, 0); font-size: 13px; font-family: HelveticaNeue, 'Helvetica Neue', Helvetica, Arial, 'Lucida Grande', sans-serif; font-style: normal; background-color: transparent;"> </div> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 10pt;" class=""> <div style="font-family: HelveticaNeue, Helvetica Neue, Helvetica, Arial, Lucida Grande, sans-serif; font-size: 12pt;" class=""> <div dir="ltr" class="" style=""> <hr size="1" class="" style=""> <font size="2" face="Arial" class="" style=""> <b class="" style=""><span style="font-weight:bold;" class="">From:</span></b> Huseyin Kaya <huseyinkaya@yahoo.com><br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">To:</span></b> Kaloyan Kovachev <kkovachev@varna.net>;
"asterisk-ss7@lists.digium.com" <asterisk-ss7@lists.digium.com> <br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">Sent:</span></b> Friday, August 22, 2014 5:32 PM<br class="" style=""> <b class="" style=""><span style="font-weight: bold;" class="">Subject:</span></b> Re: [asterisk-ss7] T38 fax issue<br class="" style=""> </font> </div> <div class="" style=""><br class="" style=""><div id="yiv1688099926" class="" style=""><div class="" style=""><table cellspacing="0" cellpadding="0" border="0" class="" style=""><tbody class="" style=""><tr class="" style=""><td colspan="1" rowspan="1" valign="top" class="" style=""><div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">Hello</font></font><br clear="none" class="" style=""></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">Actually it will be fine If I can send sip 488 not acceptable here to the re invite of t38 .</font></font></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">But Asterisk is sending sip 200 ok with voice codecs in SDP.</font></font></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">So if i was able to send sip 488 . The fax will contunie with g711 </font></font></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">Setting faxopt(gateway)=yes and t38pt_udptl = yes,redundancy,maxdatagram=400 didnt work. </font></font><br clear="none" class="" style="">
<font size="2" class="" style=""><font size="2" class="" style="">By setting these asterisk should send sip 200 with SDP T38 but it is still sending speech codec(g11u,etc..) in SIP 200 sdp.</font></font></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">Also disabling t38 with t38pt_udptl=no didnt change anything. </font></font><br clear="none" class="" style=""><br clear="none" class="" style=""></div>
<div dir="ltr" class="" style=""><font size="2" class="" style=""><font size="2" class="" style="">Regards</font></font><br clear="none" class="" style="">
</div>
</td></tr></tbody></table> <div class="" id="yiv1688099926yqt73711" style=""><div id="yiv1688099926_origMsg_" class="" style="">
<div class="" style="">
<br clear="none" class="" style="">
<div class="" style="">
<div style="font-size:0.9em;" class="">
<hr size="1" class="" style="">
<b class="" style="">
<span style="font-weight:bold;" class="">From:</span>
</b>
Kaloyan Kovachev <kkovachev@varna.net>; <br clear="none" class="" style="">
<b class="" style="">
<span style="font-weight:bold;" class="">To:</span>
</b>
Huseyin Kaya <huseyinkaya@yahoo.com>; <asterisk-ss7@lists.digium.com>; <br clear="none" class="" style="">
<b class="" style="">
<span style="font-weight:bold;" class="">Subject:</span>
</b>
Re: [asterisk-ss7] T38 fax issue <br clear="none" class="" style="">
<b class="" style="">
<span style="font-weight:bold;" class="">Sent:</span>
</b>
Fri, Aug 22, 2014 1:52:44 PM <br clear="none" class="" style="">
</div>
<br clear="none" class="" style="">
<table cellspacing="0" cellpadding="0" border="0" class="" style=""><tbody class="" style=""><tr class="" style=""><td colspan="1" rowspan="1" valign="top" class="" style="">Hi,<br clear="none" class="" style="">in addition to FAXOPT(gateway) you may try to request transmission <br clear="none" class="" style="">medium from telco - see SS7_TMR or SS7_TMR_NUM, as it should be set on <br clear="none" class="" style="">the outgoing (dahdi) channel you need to set it on the SIP channel with <br clear="none" class="" style="">underscore (_SS7_TMR)<br clear="none" class="" style=""><br clear="none" class="" style="">The possible options are defined in libss7.h and SS7_TMR_3K1_AUDIO (or 3 <br clear="none" class="" style="">as num) works fine here. If you can detect the fax calls you may request <br clear="none" class="" style="">64K_UNRESTRICTED data for them<br clear="none" class="" style=""><div class=""
id="yiv1688099926yqtfd89673" style=""><br clear="none" class="" style="">On 2014-08-22 15:47, Huseyin Kaya wrote:<br clear="none" class="" style=""><br clear="none" class="" style="">> Hello<br clear="none" class="" style="">> <br clear="none" class="" style="">> I am using Sangoma A104DE with libss7 on Asterisk 11.5.0 and <br clear="none" class="" style="">> terminating calls to telco with ss7 . We are using patched version of <br clear="none" class="" style="">> Libss7 that have timers <br clear="none" class="" style="">> functionality.(<a rel="nofollow" shape="rect" target="_blank" href="https://issues.asterisk.org/jira/browse/SS7-27" class="" style="">https://issues.asterisk.org/jira/browse/SS7-27</a>)<br clear="none" class="" style="">> <br clear="none" class="" style="">> The server is on production for the last 6 months and everything was <br clear="none" class="" style="">> fine<br clear="none" class="" style="">>
<br clear="none" class="" style="">> My interconnection with telco is only one way. From sip to ss7 .<br clear="none" class="" style="">> <br clear="none" class="" style="">> Everything is fine except one thing.<br clear="none" class="" style="">> <br clear="none" class="" style="">> One of my customers asked for t38 faxing. then i found myself in <br clear="none" class="" style="">> trouble.<br clear="none" class="" style="">> <br clear="none" class="" style="">> I get several sip traces and found the problem at the end .When i <br clear="none" class="" style="">> receive an invite T38 , asterisk is sending 200 OK message but in SDP <br clear="none" class="" style="">> it is sending g711u,g711a,g729 as codecs. So after this point
<br clear="none" class="" style="">> everything is messed up.<br clear="none" class="" style="">> <br clear="none" class="" style="">> I tried to set t38pt_udptl=no in sip.conf , but still asterisk is <br clear="none" class="" style="">> sending sip 200 ok with sdp g711...<br clear="none" class="" style="">> <br clear="none" class="" style="">> ı tried to set t38pt_udptl = yes,redundancy,maxdatagram=400 and <br clear="none" class="" style="">> setvar=FAXOPT(gateway)=yes,20 in sip.conf but still i couldn't manage <br clear="none" class="" style="">> to receive fax<br clear="none" class="" style="">> <br clear="none" class="" style="">> So basically what i need to send fax from sip to telco but could not <br clear="none" class="" style="">> manage to do till now.<br clear="none" class="" style="">> <br clear="none" class="" style="">> Everything except fax is working like a charm.<br clear="none" class=""
style="">> <br clear="none" class="" style="">> Is anyone succeed to handle fax with libss7.<br clear="none" class="" style="">> <br clear="none" class="" style="">> I will be glad if an expert can show me a way to achive this.<br clear="none" class="" style="">> <br clear="none" class="" style="">> Best Regards<br clear="none" class="" style=""></div></td></tr></tbody></table>
</div>
</div>
</div></div>
</div></div><br class="" style=""><br class="" style=""></div> </div> </div> </div></body></html>