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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'>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></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>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></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>The ACK might even be sent for real, but have the incorrect source IP so asterisk ignores it.<o:p></o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'><o:p> </o:p></span></p><p class=MsoNormal><span style='mso-fareast-language:EN-US'>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></span></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 00:54<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 7:45 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'>Hello.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I have a problem with the native Android SIP client, not acknowledging the call.<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'>Sent a message to the list for some weeks ago containing a sip debug log, but it only got stuck in moderation queue due to too large size (and it said I would get a message if moderators rejected it, but did not get message and I don’t think it got posted to list either)<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'>This ONLY happens when calling outgoing from the Android SIP client. Incoming calls works flawlessly.<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'>Everything works, audio in both directions, but the call is dropped after 30 sec.<o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>I have debugged it very much, and it seems that either Android is sending the acknowledge of the call to the incorrect IP (perhaps to the 3G network instead of via the VPN), or not sending it at all.<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'>BUT – Everything else is working flawlessly, including audio in both directions.<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'>So this means, I need somehow to tell Asterisk to ignore the lack of acknowledgement.<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'> <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'>So now to the question, since the call is dropped automatically after 30 sec with ”NO_USER_RESPONSE” (Hangupcause 18) on the far end (the callee’s end), propably because the Android native Client is not acknowledging the connected call , is it possible to tell Asterisk to just ignore the lack of acknowledgement from Android somehow?<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'>Basically, for Client sip09 (username), never hang up for the reason 18 (NO_USER_RESPONSE), threat like user response was received always.<o:p></o:p></p></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>There is no ability to ignore the lack of an ACK, as that violates the SIP standard itself.<o:p></o:p></p></div></div><div><p class=MsoNormal><o:p> </o:p></p></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>