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</o:shapelayout></xml><![endif]--></head><body lang=SV link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='mso-fareast-language:EN-US'>FINALLY solved it… Googled around for the problem, and found this:<o:p></o:p></span></p><p class=MsoNormal><a href="https://support.yeastar.com/hc/en-us/articles/360020908914-Call-Hangs-up-at-30-Seconds">https://support.yeastar.com/hc/en-us/articles/360020908914-Call-Hangs-up-at-30-Seconds</a><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Apparently, sendrpid=yes causes Android Native SIP client not to respond to the packets, and this drops the call after 30 seconds.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Disabling sendrpid makes it work successfully.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><b>Från:</b> asterisk-users <asterisk-users-bounces@lists.digium.com> <b>För </b>Joshua C. Colp<br><b>Skickat:</b> den 17 november 2019 01:18<br><b>Till:</b> Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com><br><b>Ämne:</b> Re: [asterisk-users] Disable NO_USER_RESPONSE (Hangupcause = 18) for certain SIP peer<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal>On Sat, Nov 16, 2019 at 7:59 PM Sebastian Nielsen <<a href="mailto:sebastian@sebbe.eu">sebastian@sebbe.eu</a>> wrote:<o:p></o:p></p></div><div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones?<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I do not know if the bug is in Android native SIP, or Samsung’s build of the SIP client, or if the bug is even with the OpenVPN client, or where the bug actually is.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>The ACK might even be sent for real, but have the incorrect source IP so asterisk ignores it.<o:p></o:p></p></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>The ACK is sent to the Contact header of the 200 OK sent to the phone. Using the respective logging (sip set debug on or pjsip set logger on) would tell you the IP address and port that Asterisk is telling the phone to send to, and isolate the problem further. Asterisk also doesn't ignore the ACK based on source IP address. If it shows up at Asterisk, it'll get processed.<o:p></o:p></p></div><div><p class=MsoNormal> <o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0cm 0cm 0cm 6.0pt;margin-left:4.8pt;margin-right:0cm'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'> <o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>Since audio works in both directions, it seems that the lack of ACK wouldn’t hurt (other than asterisk forcefully disconnecting the call) so I need to just tell Asterisk to not forcefully disconnect the callee.<o:p></o:p></p></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Without modifying code there's no way. The 200 OK retransmits until it gives up, and the call is disconnected.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div></div><p class=MsoNormal>-- <o:p></o:p></p><div><div><p class=MsoNormal>Joshua C. Colp<br>Digium - A Sangoma Company | Senior Software Developer<br>445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>Check us out at: <a href="http://www.sangoma.com/" target="_blank">www.sangoma.com</a> & <a href="http://www.asterisk.org/" target="_blank">www.asterisk.org</a><o:p></o:p></p></div></div></div></div></body></html>