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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple style='word-wrap:break-word'><div class=WordSection1><p class=MsoNormal>16.11.1<o:p></o:p></p><p class=MsoNormal>Works ok for me now also got already 1.3m calls on it not the last version but works fine <o:p></o:p></p><div><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span dir=LTR><o:p> </o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span lang=HE style='font-family:"Arial",sans-serif'>בברכה,</span><span dir=LTR><o:p></o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span dir=LTR><img width=253 height=89 style='width:2.6354in;height:.927in' id="תמונה_x0020_1" src="cid:image001.png@01D6C8D9.E30E9350"><o:p></o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span dir=RTL></span><span dir=RTL></span><span lang=HE style='font-size:14.0pt;font-family:"Arial",sans-serif;color:gray'><span dir=RTL></span><span dir=RTL></span> תקשורת שמדברת עסקים</span><span dir=LTR style='font-size:14.0pt;color:gray'><o:p></o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span lang=HE style='font-family:"Arial",sans-serif'><o:p> </o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><b><span lang=HE style='font-size:10.0pt;font-family:"Arial",sans-serif;color:#92D050'>שעות פעילות המשרד - 10:00-19:00</span></b><b><span lang=HE style='font-size:10.0pt;color:#92D050'><o:p></o:p></span></b></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><b><span lang=HE style='font-size:10.0pt;font-family:"Arial",sans-serif;color:#A6A6A6'>טלפון - 072-2280800<o:p></o:p></span></b></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><b><span lang=HE style='font-size:10.0pt;font-family:"Arial",sans-serif;color:#92D050'>כתובתנו באתר - <a href="http://www.phonecall.co/"><span lang=EN-US dir=LTR style='font-family:"Calibri",sans-serif;color:#92D050'>http://www.phonecall.co</span></a></span></b><span lang=HE><o:p></o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span lang=HE style='font-family:"Arial",sans-serif'><o:p> </o:p></span></p><p class=MsoNormal dir=RTL style='text-align:right;direction:rtl;unicode-bidi:embed'><span dir=LTR><o:p> </o:p></span></p></div><p class=MsoNormal><o:p> </o:p></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0cm 0cm 0cm'><p class=MsoNormal><b>From:</b> asterisk-dev <asterisk-dev-bounces@lists.digium.com> <b>On Behalf Of </b>Steve Sether<br><b>Sent:</b> Wednesday, December 2, 2020 6:14 PM<br><b>To:</b> asterisk-dev@lists.digium.com<br><b>Subject:</b> Re: [asterisk-dev] PJSIP on Asterisk 16 doesn't send t38 packets when sending via res_fax module.<o:p></o:p></p></div></div><p class=MsoNormal><o:p> </o:p></p><pre>This is in fact the exact problem. We use local channels when sending out faxes.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>Thanks for your quick reply.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>On Tue, Dec 1, 2020 at 6:23 PM Steve Sether <<a href="http://lists.digium.com/mailman/listinfo/asterisk-dev">ssether at usinternet.com</a>> wrote:<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>><i> I'm having a lot of trouble getting PJSIP to send a fax via T.38 in<o:p></o:p></i></pre><pre>><i> Asterisk 16.14.1.<o:p></o:p></i></pre><pre>><i><o:p> </o:p></i></pre><pre>><i> From the pcap, I see the call being made, it connects, RTP goes in both<o:p></o:p></i></pre><pre>><i> directions, the far-end requests a re-invite using t.38, Asterisk sends<o:p></o:p></i></pre><pre>><i> an OK, the OK is acked, and the far ends starts sending t.38 packets,<o:p></o:p></i></pre><pre>><i> asterisk stops sending RTP.<o:p></o:p></i></pre><pre>><i><o:p> </o:p></i></pre><pre>><i> Then Asterisk basically sends nothing else. No T.38 packets are sent by<o:p></o:p></i></pre><pre>><i> Asterisk, No SIP, nothing. The far end eventually gives up sending<o:p></o:p></i></pre><pre>><i> t.38, and sends a bye, ending the call.<o:p></o:p></i></pre><pre>><i><o:p> </o:p></i></pre><pre><o:p> </o:p></pre><pre>Do you have a Local channel in the mix? If so then there is a known<o:p></o:p></pre><pre>issue[1] which is being worked on.<o:p></o:p></pre><pre><o:p> </o:p></pre><pre>[1] <a href="https://issues.asterisk.org/jira/browse/ASTERISK-29035">https://issues.asterisk.org/jira/browse/ASTERISK-29035</a><o:p></o:p></pre><pre><o:p> </o:p></pre><pre>-- <o:p></o:p></pre><pre>Joshua C. Colp<o:p></o:p></pre><pre>Asterisk Technical Lead<o:p></o:p></pre><pre>Sangoma Technologies<o:p></o:p></pre><pre>Check us out at <a href="http://www.sangoma.com">www.sangoma.com</a> and <a href="http://www.asterisk.org">www.asterisk.org</a><o:p></o:p></pre><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><p class=MsoNormal style='margin-bottom:12.0pt'><br><br><o:p></o:p></p></blockquote><div><p class=MsoNormal>-- <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p></div></div></body></html>