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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'>A quick glance at your logs tells me that your SIP provider/endpoint at 66.8.50.218:5060 is rejecting the call <a href="sip:**********@sip.*****.co.za%20SIP/2.0">sip:**********@sip.*****.co.za SIP/2.0</a>(line 46) by saying: “SIP/2.0 404 Not Found”<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'>Although the number is filtered out for privacy reasons, I would double check that the dialplan at endpoint at 66.8.50.218 will accept the number referenced in sip: <b><u>**********</u> </b>@ sip.*****.co.za because that end point is actively rejecting that number.<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 haven’t seen a post to this mailing list for a long time. You may get better traction in the asterisk-users or in IRC as this seems to be more related to standard SIP configuration.<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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'><o:p> </o:p></span></b></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:#1F497D'>Karl Schmutz<br></span></b><b><i><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:maroon'>Networking Systems Engineer<br></span></i></b><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:navy'>Starnet Data Design, Inc.</span></b><span style='color:#1F497D'><br></span><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:navy'>Direct Line: 805.277.0117<br>Toll Free: 800.779.0587<br>Westlake Village, CA - Phoenix, AZ<br></span><i><u><span style='font-size:10.0pt;font-family:"Franklin Gothic Book","sans-serif";color:blue'><a href="http://www.starnetdata.com">www.starnetdata.com</a></span></u></i><i><span style='color:#1F497D'><br></span></i><b><span style='font-size:10.0pt;font-family:"Verdana","sans-serif";color:lime;background:blue'>Service. Value. Integrity.<br></span></b><b><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:navy'>Follow Us on Twitter: </span></b><span style='font-size:7.5pt;font-family:"Verdana","sans-serif";color:navy'><a href="http://www.twitter.com/Starnet_Data"><span style='color:blue'>www.twitter.com/Starnet_Data</span></a></span><span style='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-embedded-bounces@lists.digium.com [mailto:asterisk-embedded-bounces@lists.digium.com] <b>On Behalf Of </b>helge.reikeras@gmail.com<br><b>Sent:</b> Thursday, May 12, 2011 7:52 AM<br><b>To:</b> asterisk-embedded@lists.digium.com<br><b>Subject:</b> [asterisk-embedded] Problem with PSTN calls (Asterisk as SIP clienton embedded device)<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Hi <o:p></o:p></p><div><p class=MsoNormal>I've spent two days trying to solve this issue but to no prevail and I'm hoping to get some help.<o:p></o:p></p></div><div><p class=MsoNormal><br>I've configured Asterisk as a SIP client, running on OpenWRT on an embedded device with onboard FXS and ATA. Asterisk is connecting to an external SIP provider on the Internet who in turn provides a PSTN gateway. I'm able to make calls to other SIP accounts registered on the same server who are outside my LAN. However, I can not make calls to any PSTN numbers. When trying to make PSTN calls it sounds like the person at the other end is immediately rejecting the call although I know this is not the case.<br><br>Firstly, I'm absolutely sure that the PSTN gateway is working because I can make outbound PSTN calls with the same SIP account using other SIP clients (Empathy-SIP, SIPDroid) from the same LAN. However, when registering the same SIP account using Asterisk from OpenWRT all PSTN calls fail. Inbound calls from PSTN numbers also fail while calls from other SIP clients on the same server work fine. The SIP accounts shows as registered in Asterisk.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal><span style='font-family:"Arial","sans-serif"'>I've attached detailed error logs. The log files 'messages-pstn.log' shows the failed (PSTN) call and 'messages-voip.log' shows the successful (VOIP) call. Note that I have replaced actual phone numbers and domain names with *** for anonymity.</span><br><span style='font-family:"Arial","sans-serif"'><br>I suspect perhaps a codec issue, but I haven't been able to identify the actual problem. Any ideas that will help me towards solving this problem is greatly appreciated.<br></span><br><span style='font-family:"Arial","sans-serif"'>Regards,</span><br><span style='color:#888888'>Helge</span><o:p></o:p></p></div></div></div></div></body></html>