<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div><br></div><div>Scott, thank you four your reply.</div><div><br></div><div>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.</div><div><br></div><div>Only after a sip reload the peer works again.</div><div><br></div><div>That can't be normal...</div><div><br></div><div><div>Daniel</div></div><div><br>Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <<a href="mailto:sgriepentrog@digium.com">sgriepentrog@digium.com</a>>:<br><br></div><blockquote type="cite"><div><div dir="ltr"><div class="gmail_default" style="color:#660000">You have two options for dealing with an IP change during the registration period:</div><div class="gmail_default" style="color:#660000"><br></div><div class="gmail_default" style="color:#660000">1) set the registration time to shorter period of time to minimize the downtime</div><div class="gmail_default" style="color:#660000"><br></div><div class="gmail_default" 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.</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl <span dir="ltr"><<a href="mailto:daniel.heckl@gmail.com" target="_blank">daniel.heckl@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="auto"><div><span style="background-color:rgba(255,255,255,0)">Maybe someone could elaborate on my first question again.</span></div><div><span style="background-color:rgba(255,255,255,0)"><br></span></div><div>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?</div><div><div class="h5"><div><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>>:<br><br></div><blockquote type="cite"><div>Hello Sebastian,<div><br></div><div>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.</div><div><br></div><div>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.</div><div><br></div><div>If I change insecure to insecure=port,invite - could that be a solution?</div><div><br></div><div>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?</div><div><div>
<div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br></div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px">Daniel</div><div style="color:rgb(0,0,0);font-family:Helvetica;font-size:12px;font-style:normal;font-variant:normal;font-weight:normal;letter-spacing:normal;line-height:normal;text-align:start;text-indent:0px;text-transform:none;white-space:normal;word-spacing:0px"><br></div></div><div><blockquote type="cite"><div>Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <<a href="mailto:sebastian_ml@gmx.net" target="_blank">sebastian_ml@gmx.net</a>>:</div><br><div>On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:<br><blockquote type="cite">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.<br></blockquote><br>Hello Daniel,<br><br>I'll find myself in the same situation a few weeks from now :-)<br><br><blockquote type="cite"><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?<br></blockquote><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><blockquote type="cite">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<br></blockquote><br><blockquote type="cite">-- <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? 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</div></blockquote><blockquote type="cite"><div><span>-- </span><br><span>_____________________________________________________________________</span><br><span>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --</span><br><span>New to Asterisk? Join us for a live introductory webinar every Thurs:</span><br><span> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a></span><br><span></span><br><span>asterisk-users mailing list</span><br><span>To UNSUBSCRIBE or update options visit:</span><br><span> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a></span></div></blockquote></body></html>