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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'> If I correctly understand what the problem is, what I did was write a script that runs out of CRON every 15 minutes. It checks the outside IP address by querying <a href="http://checkip.dyndns.org">http://checkip.dyndns.org</a> and compares it to the IP address stored in the parameter “externip” in the [general] section of sip.conf. If the two values are the same, the script exits quietly. If they are different, the script updates “externip” with the new address, does a sip reload, and shoots me an email saying there was an update. It's a fairly simple and straightforward process and does the job. I use this script for all PBX’s that are behind a NAT. I hope this helps.<o:p></o:p></span></p><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>Regards;<o:p></o:p></span></p><p class=MsoNormal style='text-align:justify'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>John<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><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><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>Daniel Heckl<br><b>Sent:</b> Wednesday, April 01, 2015 2:56 AM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b>Subject:</b> Re: [asterisk-users] Update peer IP address<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Scott, thank you four your reply.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I had already though about both options, but the problem is, that after an ip change AND a new registration the ip address of the peer is not updated automatically. INVITES are answered with 401.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Only after a sip reload the peer works again.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>That can't be normal...<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><div><p class=MsoNormal>Daniel<o:p></o:p></p></div></div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <<a href="mailto:sgriepentrog@digium.com">sgriepentrog@digium.com</a>>:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><div><p class=MsoNormal><span style='color:#660000'>You have two options for dealing with an IP change during the registration period:<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='color:#660000'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='color:#660000'>1) set the registration time to shorter period of time to minimize the downtime<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='color:#660000'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='color:#660000'>2) detect that the IP address has changed via whatever method available, and then issue a "sip reload" CLI command to asterisk, which will cause it to resend registrations immediately.<o:p></o:p></span></p></div></div><div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl <<a href="mailto:daniel.heckl@gmail.com" target="_blank">daniel.heckl@gmail.com</a>> wrote:<o:p></o:p></p><div><div><p class=MsoNormal>Maybe someone could elaborate on my first question again.<o:p></o:p></p></div><div><p class=MsoNormal><br><br><o:p></o:p></p></div><div><p class=MsoNormal>If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer?<o:p></o:p></p></div><div><div><div><p class=MsoNormal style='margin-bottom:12.0pt'><br>Am 31.03.2015 um 12:36 schrieb Daniel Heckl <<a href="mailto:daniel.heckl@gmail.com" target="_blank">daniel.heckl@gmail.com</a>>:<o:p></o:p></p></div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal>Hello Sebastian,<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>I had already seen this list of the hosts, but it is not active. All servers with which my Asterisk has been communicated are not listed.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>A port scan, to eventually update the list, found hundreds of servers provided in the address range <a href="http://217.0.0.0/13" target="_blank">217.0.0.0/13</a> with open port 5060, some were even not found. I think there must be another solution.<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>If I change insecure to insecure=port,invite - could that be a solution?<o:p></o:p></p></div><div><p class=MsoNormal><o:p> </o:p></p></div><div><p class=MsoNormal>Or should I try to change to res_pjsip (upgrade to Asterisk 13 is no problem)? Has there anyone experience with dynamic ip addresses of Asterisk?<o:p></o:p></p></div><div><div><div><p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'><o:p> </o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'>Daniel<o:p></o:p></span></p></div><div><p class=MsoNormal><span style='font-size:9.0pt;font-family:"Helvetica","sans-serif";color:black'><o:p> </o:p></span></p></div></div><div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal>Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <<a href="mailto:sebastian_ml@gmx.net" target="_blank">sebastian_ml@gmx.net</a>>:<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:<br><br><o:p></o:p></p><p class=MsoNormal>Hello<br><br>I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom<br>Germany. We have sometimes problems with incoming and outgoing calls.<br>I hope I can explain it understandable.<o:p></o:p></p><p class=MsoNormal><br>Hello Daniel,<br><br>I'll find myself in the same situation a few weeks from now :-)<br><br><br><o:p></o:p></p><p class=MsoNormal><br>For example, Asterisk sends a REGISTER to 217.0.23.68 (<a href="http://tel.t-online.de" target="_blank">tel.t-online.de</a><br><<a href="http://tel.t-online.de/" target="_blank">http://tel.t-online.de/</a>>), the message is answered with OK and the<br>peer is registered.<br><br>Usually INVITES comes now from this ip address. All works fine. But<br>sometimes INVITES comes from an other IP address, for example<br>217.0.23.100. This request Asterisk responds with 401 Unauthorized.<br><br>In the next register procedure REGISTER are sent to the new ip address<br>and answered also with OK. But qualify OPTIONS are continue be sent to<br>the old ip address. Incoming and outgoing calls are canceled. Outgoing<br>calls are answered with Forbidden.<br><br>Even if the REGISTER procedure works with the new ip address, the<br>peers are connected with the old address.<br><br>Waiting doesn’t help, only a „sip reload“ update the ip address of the<br>peer. <br><br>What is the solution for this problem? How can asterisk update the<br>peer?<o:p></o:p></p><p class=MsoNormal><br>I think the solution - for the inbound issue at least - could be to add<br>more hosts as a peer. Have a looks at this forum post:<br><br><a href="http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371" target="_blank">http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371</a><br><br>The user used a template and than he added peers, each with its own IP<br>address. The provided list was last updated in 2014, though, so I assume<br>the provider in the meantime has added to that list.<br><br>It looks pretty tedious, though, I mean there could be dozens of IPs<br>you'd have to add. But I guess this is the way to go with Asterisk 11<br>and chan_sip.<br><br>The future looks brighter :-) I read that with pjsip, which I understand<br>is the replacement for chan_sip, you can have one peer entry and match<br>an IP range instead of a single host. That should tidy up the dialplan.<br><br>What I'm a little afraid of is the SIP provider using IPs out of a range<br>that they also use for other services. Maybe out of the same range they<br>hand out IPs to their customers. I guess we got to be careful :-)<br><br>Kind regards,<br>Sebastian<br><br><br><o:p></o:p></p><p class=MsoNormal>The Asterisk is local behind a NAT with a firewall, following settings<br>are used:<br><br>externhost with DynDNS stun with <a href="http://stun.t-online.de" target="_blank">stun.t-online.de</a><br><<a href="http://stun.t-online.de/" target="_blank">http://stun.t-online.de/</a>> nat=yes srvlookup=yes allowguest=no<br>trustrpid=no insecure=invite qualify=yes<br><br>Thank you! Daniel<o:p></o:p></p><p class=MsoNormal><br><br><o:p></o:p></p><p class=MsoNormal>-- <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><p class=MsoNormal><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><o:p></o:p></p></div></blockquote></div><p class=MsoNormal><o:p> </o:p></p></div></div></blockquote></div></div></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></div><p class=MsoNormal><br><br clear=all><o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>-- <o:p></o:p></p><div><div><p class=MsoNormal><span style='border:solid windowtext 1.0pt;padding:0in'><img border=0 width=288 height=50 id="_x0000_i1025" src="cid:image001.jpg@01D06C67.03FCB9F0" alt="Image removed by sender. Digium logo"></span><o:p></o:p></p><div><p class=MsoNormal>Scott Griepentrog<br>Digium, Inc · Software Developer<br>445 Jan Davis Drive NW · Huntsville, AL 35806 · US<br>direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090<br>Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> · <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><o:p></o:p></p></div></div></div></div></div></blockquote><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><p class=MsoNormal>-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">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">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">http://lists.digium.com/mailman/listinfo/asterisk-users</a><o:p></o:p></p></div></blockquote></div></body></html>