<html><head><meta http-equiv="Content-Type" content="text/html charset=utf-8"></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space;" class="">Hello Sebastian,<div class=""><br class=""></div><div class="">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 class=""><br class=""></div><div class="">A port scan, to eventually update the list, found hundreds of servers provided in the address range 217.0.0.0/13 with open port 5060, some were even not found. I think there must be another solution.</div><div class=""><br class=""></div><div class="">If I change insecure to insecure=port,invite - could that be a solution?</div><div class=""><br class=""></div><div class="">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 class=""><div apple-content-edited="true" class="">
<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; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" class=""><br class="Apple-interchange-newline"></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; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" class="">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; orphans: auto; text-align: start; text-indent: 0px; text-transform: none; white-space: normal; widows: auto; word-spacing: 0px; -webkit-text-stroke-width: 0px;" class=""><br class=""></div></div><div><blockquote type="cite" class=""><div class="">Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <<a href="mailto:sebastian_ml@gmx.net" class="">sebastian_ml@gmx.net</a>>:</div><br class="Apple-interchange-newline"><div class="">On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote:<br class=""><blockquote type="cite" class="">Hello<br class=""><br class="">I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom<br class="">Germany. We have sometimes problems with incoming and outgoing calls.<br class="">I hope I can explain it understandable.<br class=""></blockquote><br class="">Hello Daniel,<br class=""><br class="">I'll find myself in the same situation a few weeks from now :-)<br class=""><br class=""><blockquote type="cite" class=""><br class="">For example, Asterisk sends a REGISTER to 217.0.23.68 (<a href="http://tel.t-online.de" class="">tel.t-online.de</a><br class=""><<a href="http://tel.t-online.de/" class="">http://tel.t-online.de/</a>>), the message is answered with OK and the<br class="">peer is registered.<br class=""><br class="">Usually INVITES comes now from this ip address. All works fine. But<br class="">sometimes INVITES comes from an other IP address, for example<br class="">217.0.23.100. This request Asterisk responds with 401 Unauthorized.<br class=""><br class="">In the next register procedure REGISTER are sent to the new ip address<br class="">and answered also with OK. But qualify OPTIONS are continue be sent to<br class="">the old ip address. Incoming and outgoing calls are canceled. Outgoing<br class="">calls are answered with Forbidden.<br class=""><br class="">Even if the REGISTER procedure works with the new ip address, the<br class="">peers are connected with the old address.<br class=""><br class="">Waiting doesn’t help, only a „sip reload“ update the ip address of the<br class="">peer. <br class=""><br class="">What is the solution for this problem? How can asterisk update the<br class="">peer?<br class=""></blockquote><br class="">I think the solution - for the inbound issue at least - could be to add<br class="">more hosts as a peer. Have a looks at this forum post:<br class=""><br class=""><a href="http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371" class="">http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371</a><br class=""><br class="">The user used a template and than he added peers, each with its own IP<br class="">address. The provided list was last updated in 2014, though, so I assume<br class="">the provider in the meantime has added to that list.<br class=""><br class="">It looks pretty tedious, though, I mean there could be dozens of IPs<br class="">you'd have to add. But I guess this is the way to go with Asterisk 11<br class="">and chan_sip.<br class=""><br class="">The future looks brighter :-) I read that with pjsip, which I understand<br class="">is the replacement for chan_sip, you can have one peer entry and match<br class="">an IP range instead of a single host. That should tidy up the dialplan.<br class=""><br class="">What I'm a little afraid of is the SIP provider using IPs out of a range<br class="">that they also use for other services. Maybe out of the same range they<br class="">hand out IPs to their customers. I guess we got to be careful :-)<br class=""><br class="">Kind regards,<br class="">Sebastian<br class=""><br class=""><blockquote type="cite" class="">The Asterisk is local behind a NAT with a firewall, following settings<br class="">are used:<br class=""><br class="">externhost with DynDNS stun with stun.t-online.de<br class=""><http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no<br class="">trustrpid=no insecure=invite qualify=yes<br class=""><br class="">Thank you! Daniel<br class=""></blockquote><br class=""><blockquote type="cite" class="">-- <br class="">_____________________________________________________________________<br class="">-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br class="">New to Asterisk? Join us for a live introductory webinar every Thurs:<br class=""> http://www.asterisk.org/hello<br class=""><br class="">asterisk-users mailing list<br class="">To UNSUBSCRIBE or update options visit:<br class=""> http://lists.digium.com/mailman/listinfo/asterisk-users<br class=""></blockquote><br class=""><br class="">-- <br class="">_____________________________________________________________________<br class="">-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br class="">New to Asterisk? Join us for a live introductory webinar every Thurs:<br class=""> http://www.asterisk.org/hello<br class=""><br class="">asterisk-users mailing list<br class="">To UNSUBSCRIBE or update options visit:<br class=""> http://lists.digium.com/mailman/listinfo/asterisk-users</div></blockquote></div><br class=""></div></body></html>