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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'>Thanks George.<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'>That was the ip address I was given. Unfortunately, my contact at Vitelity is gone for the day so I can’t verify it with him.<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 added the qualify_frequency as you suggested and it does appear that I have something configured incorrectly….<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'><--- Transmitting SIP request (483 bytes) to UDP:0.0.19.196:5060 ---><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>OPTIONS sip:64.2.142.93@5060 SIP/2.0<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Via: SIP/2.0/UDP xxx.xxx.xx.xxx:5060;rport;branch=z9hG4bKPjcea63914-b8d1-483d-96db-11968abab704<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>From: <sip:e31d5809-f26a-4219-8365-70931428072b@xxx.xxx.xx.xxx>;tag=7cfab3ba-73de-4243-9967-d1e6a5e7b0b4<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>To: <sip:64.2.142.93@5060><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Contact: <sip:e31d5809-f26a-4219-8365-70931428072b@xxx.xxx.xx.xxx:5060><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Call-ID: 7ba766bf-363b-47d0-a388-62a58d1df88d<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>CSeq: 33778 OPTIONS<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Max-Forwards: 70<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>User-Agent: Asterisk PBX 13.0.0<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Content-Length: 0<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'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>[Dec 17 19:22:31] WARNING[49476]: pjsip:0 <?>: tsx0x3c501e8 .Failed to send Request msg OPTIONS/cseq=33778 (tdta0x32c7c90)! err=120022 (Invalid argument)<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>[Dec 17 19:22:31] ERROR[49476]: res_pjsip.c:2532 endpt_send_request: Error 120022 'Invalid argument' sending OPTIONS request to endpoint <unknown><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'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>The 64.2.142.93 is the exact value I was given to use for the outbound trunk (works with sip.conf)<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'>host=64.2.142.93<o:p></o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Any thoughts?<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I was really hoping they had worked with the PJSIP, but apparently the latest Asterisk version any of their customers are using is Asterisk 11.<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'>Have a great day!<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'>Dan<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> Wednesday, December 10, 2014 2:40 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><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Wed, Dec 10, 2014 at 1:27 PM, Dan Cropp <<a href="mailto:dan@amtelco.com" target="_blank">dan@amtelco.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me.<br><br>type=friend<br>dtmfmode=auto<br>host=64.2.142.93<br>allow=all<br>nat=yes<br>canreinvite=no<br>trustrpid=yes<br>sendrpid=yes<br><br>When I use these settings to originate calls using the sip.conf they sent me, everything works.<br><br>Action: Originate<br>ActionID: S8<br>Channel: SIP/<a href="http://outbound.vitelity.net/8005555555" target="_blank">outbound.vitelity.net/8005555555</a><br>Exten: createcall<br>Context: TestApp<br>Priority: 1<br>Timeout: 60000<br>CallerID: John Doe <1234><br>Variable: CALLERID(num-pres)=allowed_passed_screened<br>Async: true<br><br><br>I translated those settings to the following for pjsip.conf...<br><br>[transport1]<br>type = transport<br>bind = 0.0.0.0<br>protocol = udp<br><br>[<a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a>]<br>type = aor<br>remove_existing = yes<br>contact = <a href="sip:64.2.142.93@5060">sip:64.2.142.93@5060</a><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>You might want to set a qualify_frequency here to see if the server responds to OPTIONS messages. Also 64.2.142.93 isn't currently one of their outbound servers. Are you using one of their inbound* servers as outbound? IIRC unless you ask them, they don't allow it.<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'><p class=MsoNormal><br>[<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>allow = all<br>direct_media = no<br><br>[identify1]<br>type = identify<br>endpoint = <a href="http://outbound.vitelity.net" target="_blank">outbound.vitelity.net</a><br>match = 64.2.142.93<br><br>When I attempt to use AMI Originate, it's failing. I am not seeing anything with pjsip logging turned on, so it seems to be something with the settings.<br><br>Action: Originate<br>ActionID: S8<br>Channel: PJSIP/<a href="http://outbound.vitelity.net/8005555555" target="_blank">outbound.vitelity.net/8005555555</a><br>Exten: createcall<br>Context: TestApp<br>Priority: 1<br>Timeout: 60000<br>CallerID: John Doe <1234><br>Variable: CALLERID(num-pres)=allowed_passed_screened<br>Async: true<br><br>NOTE: I am able to use AMI Originate to other PJSIP endpoints.<br><br>Action: Originate<br>ActionID: S9<br>Channel: PJSIP/1003/1003<br>Exten: createcall<br>Context: TestApp<br>Priority: 1<br>Timeout: 60000<br>CallerID: John Doe <1234><br>Variable: CALLERID(num-pres)=allowed_passed_screened<br>Async: true<br><br>Anyone have any suggestions as to what I am doing wrong?<br><br>Have a great day!<br><span style='color:#888888'><br>Dan<br><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></span><o:p></o:p></p></blockquote></div><p class=MsoNormal><o:p> </o:p></p></div></div></div></body></html>