<div dir="ltr"><div dir="ltr">On Wed, Feb 3, 2021 at 11:02 PM Ruisheng Peng <<a href="mailto:rpeng@ifa.hawaii.edu">rpeng@ifa.hawaii.edu</a>> wrote:</div><div dir="ltr"><br></div><div><snip></div><div dir="ltr"><br></div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div style="font-size:small">When using handsets with udp or tcp transports to dial ext 100, it'd hangup after the no-one-arround message. However, when using the handset with tls transport, it doesn't hang up on its own if ext 100 is not answered. I have to click the hangup button to accomplish that. Here's what asterisk log shows:</div><div style="font-size:small"><br></div><p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> == Setting global variable 'SIPDOMAIN' to '128.171.77.23'</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [100@sets:1] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Dial</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/SOFTPHONE_B-00000007</span><span style="font-variant-ligatures:no-common-ligatures">", "</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/0000f30A0A01,10,m</span><span style="font-variant-ligatures:no-common-ligatures">") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- Called PJSIP/0000f30A0A01</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- Started music on hold, class 'default', on channel 'PJSIP/SOFTPHONE_B-00000007'</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> > 0x7f0fa801ede0 -- Strict RTP learning after remote address set to: <a href="http://128.171.168.233:7078" target="_blank">128.171.168.233:7078</a></font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- PJSIP/0000f30A0A01-00000008 is ringing</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- PJSIP/0000f30A0A01-00000008 is ringing</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> > 0x7f0fa801ede0 -- Strict RTP switching to RTP target address <a href="http://128.171.168.233:7078" target="_blank">128.171.168.233:7078</a> as source</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> > 0x7f0fa801ede0 -- Strict RTP learning complete - Locking on source address <a href="http://128.171.168.233:7078" target="_blank">128.171.168.233:7078</a></font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- Nobody picked up in 10000 ms</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- Stopped music on hold on PJSIP/SOFTPHONE_B-00000007</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [100@sets:2] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Playback</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/SOFTPHONE_B-00000007</span><span style="font-variant-ligatures:no-common-ligatures">", "</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">vm-nobodyavail</span><span style="font-variant-ligatures:no-common-ligatures">") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- <PJSIP/SOFTPHONE_B-00000007> Playing 'vm-nobodyavail.slin' (language 'en')</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [100@sets:3] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Hangup</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/SOFTPHONE_B-00000007</span><span style="font-variant-ligatures:no-common-ligatures">", "") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> == Spawn extension (sets, 100, 3) exited non-zero on 'PJSIP/SOFTPHONE_B-00000007'</font></span></p>
<div><font face="monospace"><span style="color:rgb(0,0,0)">voip1*CLI></span><span style="color:rgb(0,0,0)"> </span></font> </div><div><br></div><div> Another quirk is when I use a phone with udp transport (RP_Yealink) to call a phone with tls transport (RP_OMBP) it immediately jumps the no-one-around message w/o ringing, then hang up. The tls phone is shown available but asterisk sees it busy:</div><div><br></div><div><p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> == Setting global variable 'SIPDOMAIN' to '128.171.77.23'</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [103@sets:1] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Dial</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/0000f30A0A01-0000000d</span><span style="font-variant-ligatures:no-common-ligatures">", "</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/SOFTPHONE_B,10</span><span style="font-variant-ligatures:no-common-ligatures">") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- Called PJSIP/SOFTPHONE_B</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> == Everyone is busy/congested at this time (1:0/1/0)</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [103@sets:2] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Playback</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/0000f30A0A01-0000000d</span><span style="font-variant-ligatures:no-common-ligatures">", "</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">vm-nobodyavail</span><span style="font-variant-ligatures:no-common-ligatures">") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> > 0x7f0fa000c330 -- Strict RTP learning after remote address set to: <a href="http://128.171.77.118:11790" target="_blank">128.171.77.118:11790</a></font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> > 0x7f0fa000c330 -- Strict RTP switching to RTP target address <a href="http://128.171.77.118:11790" target="_blank">128.171.77.118:11790</a> as source</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> -- <PJSIP/0000f30A0A01-0000000d> Playing 'vm-nobodyavail.slin' (language 'en')</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><font face="monospace"><span style="font-variant-ligatures:no-common-ligatures"> -- Executing [103@sets:3] </span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(46,174,187)">Hangup</span><span style="font-variant-ligatures:no-common-ligatures">("</span><span style="font-variant-ligatures:no-common-ligatures;color:rgb(200,20,201)">PJSIP/0000f30A0A01-0000000d</span><span style="font-variant-ligatures:no-common-ligatures">", "") in new stack</span></font></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace"> == Spawn extension (sets, 103, 3) exited non-zero on 'PJSIP/0000f30A0A01-0000000d'</font></span></p>
<p style="margin:0px;font-stretch:normal;line-height:normal;color:rgb(0,0,0)"><span style="font-variant-ligatures:no-common-ligatures"><font face="monospace">voip1*CLI></font></span></p></div><div> </div><div> Suppose it's not cool to mix transports among your handsets? <font face="arial, sans-serif">Any suggestions?<br></font></div></div></div></div></div></div></div></blockquote><div><br></div><div>I'd suggest looking at the actual SIP signaling to see what is going on using "pjsip set logger on" and also providing configuration. This would allow better insight into what exactly is going on. </div></div><div><br></div>-- <br><div dir="ltr" class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr"><div dir="ltr"><div dir="ltr"><div dir="ltr"><div style="font-family:tahoma,sans-serif"><div><font color="#073763">Joshua C. Colp</font></div><div><font color="#073763">Asterisk Technical Lead</font></div><div><font color="#073763">Sangoma Technologies</font></div><div><font color="#073763">Check us out at <a href="http://www.sangoma.com/" target="_blank">www.sangoma.com</a> and <a href="http://www.asterisk.org/" target="_blank">www.asterisk.org</a></font></div></div></div></div></div></div></div></div></div></div></div></div>