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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I corrected my local_net setting (based on advice from network admin).<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I have tried several different values for the from_user and still have the same problem.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Asterisk receives the OK from Vitelity.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Asterisk sends the ACK (without a Contact header).<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Vitelity doesn’t seem to process it, so they send an OK again.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The OK receive, Transmit ACK occurs 4 times.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>A short while later, Vitelity hangs up on my cell phone.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Asterisk is never told the call is gone.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>If I hangup the call from Asterisk side,<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Asterisk sends the BYE message.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Vitelity responds with a “SIP/2.0 481 Call leg/transaction does not exist”<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Again, the trace indicates the ACK message is missing the Contact header.<o:p></o:p></span></p><p class=MsoNormal style='text-autospace:none'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal style='text-autospace:none'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Additional note: the network admin is asking why the local_net, external_media_address, and external_signalling_address are needed. He wrote me…“</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>You should NOT have to know your public IP. The firewall should be doing fixup commands to change the values in the packet”</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal style='text-autospace:none'><span style='font-size:9.0pt;font-family:"Arial","sans-serif";color:black'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>George Joseph<br><b>Sent:</b> Monday, December 15, 2014 11:14 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] PJSIP configuration question<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><div><p class=MsoNormal>On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <<a href="mailto:dan@amtelco.com" target="_blank">dan@amtelco.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal>Thanks George.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I will correct my local_net in the morning.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Vitelity chan_sip settings I have working, do not have a fromuser.<o:p></o:p></p></div><div><p class=MsoNormal>sip.conf settings...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div></div><div><p class=MsoNormal>I think you can actually specify anything, it just has to be populated with something other than a sub-account username.<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><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal>[HVout]<o:p></o:p></p><p class=MsoNormal>type=friend<o:p></o:p></p><p class=MsoNormal>dtmfmode=auto<o:p></o:p></p><p class=MsoNormal>host=64.2.142.93<o:p></o:p></p><p class=MsoNormal>disallow=all<o:p></o:p></p><p class=MsoNormal>allow=ulaw<o:p></o:p></p><p class=MsoNormal>canreinvite=no<o:p></o:p></p><p class=MsoNormal>trustrpid=yes<o:p></o:p></p><p class=MsoNormal>sendrpid=yes<o:p></o:p></p><p class=MsoNormal>nat=yes<o:p></o:p></p><p class=MsoNormal>context=TestApp<o:p></o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'><o:p> </o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>On Dec 15, 2014, at 9:32 PM, George Joseph <<a href="mailto:george.joseph@fairview5.com" target="_blank">george.joseph@fairview5.com</a>> wrote:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <<a href="mailto:dan@amtelco.com" target="_blank">dan@amtelco.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I am not sure if I entered the correct settings for the transport information.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>For the local_net, I entered my local ip address, but no mask. I will check with the network admin so he can verify the settings I entered.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p></div></div><div><p class=MsoNormal>You need the network and mask. For example if the ip address and mask of the test machine is <a href="http://192.168.0.1/255.255.255.0" target="_blank">192.168.0.1/255.255.255.0</a> then the correct entry would be <a href="http://192.168.0.0/24" target="_blank">192.168.0.0/24</a>.<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:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>One minor detail, we are using ip authentication. When Vitelity changed my account from user based authentication to IP based authentication, they stopped including a user for the account.</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Should these settings work without the from_user (IP based authentication) or do I need to get the account name from Vitelity?</span><o:p></o:p></p></div></div></blockquote><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>You definitely need the master account login username. If you has this working with chan_sip, then try the 'fromuser' from sip.conf and user is from_user.<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> <o:p></o:p></p></div><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Have a great day!</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Da</span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> </span><o:p></o:p></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>] <b>On Behalf Of </b>George Joseph<br><b>Sent:</b> Monday, December 15, 2014 7:27 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] PJSIP configuration question</span><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><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>Ok Dan, try this... I was able to get this to work behind a NAT and with ip address authentication.<br><br>[global]<br>type = global<br>debug = yes<o:p></o:p></p><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>[transport1]<br>type = transport<br>bind = 0.0.0.0<br>protocol = udp<br><b>local_net=<yourlocalnet I.E. <a href="http://10.10.10.10/24" target="_blank">10.10.10.10/24</a>><br>external_media_address=<your public ip address><br>external_signaling_address=<your public address><br></b> <br>[<a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a>]<br>type = aor<br>remove_existing = yes<br>qualify_frequency = 60<br>contact = <a href="sip:64.2.142.93">sip:64.2.142.93</a><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>[<a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a>]<br>type = endpoint<br>context = TestApp<br>transport = transport1<br>aors = <a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a><br>dtmf_mode = rfc4733<br>force_rport = yes<br>rtp_symmetric = yes<br>rewrite_contact = yes<br>send_rpid = yes<br>trust_id_inbound = yes<br>disallow = all<br>allow = ulaw<br>direct_media = no<o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'><b>from_user=<your main vitelity account name> ; Not subaccount</b><o:p></o:p></p></div><div><p class=MsoNormal style='mso-margin-top-alt:auto;margin-bottom:12.0pt'>[<a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a>]<br>type = identify<br>endpoint = <a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a><br>match = 64.2.142.93<o:p></o:p></p></div></div></div></div><p class=MsoNormal><br><span class=hoenzb><span style='color:#888888'>--</span></span><span style='color:#888888'><br><span class=hoenzb>_____________________________________________________________________</span><br><span class=hoenzb>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --</span><br><span class=hoenzb>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span class=hoenzb> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a></span><br><br><span class=hoenzb>asterisk-users mailing list</span><br><span class=hoenzb>To UNSUBSCRIBE or update options visit:</span><br><span class=hoenzb> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></span><o:p></o:p></p></blockquote></div></div></div></div></blockquote><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal><span style='color:#888888'>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></span></p></div></blockquote></div><p class=MsoNormal><br>--<br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>New to Asterisk? Join us for a live introductory webinar every Thurs:<br> <a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></blockquote></div></div></div></div></body></html>