Actually, the IP address is still used to identify the incoming invite. With the insecure=port option set, Asterisk will presume the invite to still match the trunk account even if the NAT router has mangled (changed) the port number. My suspicion is that when the new register goes out, it's creating a new state in the firewall, resulting in a new port number, which is why you would have to allow anonymous calls to then accept it without insecure=port. The other possibility is that you have a port forward in the router set, which is similarly mangling the port number. With a valid registration being held, and assuming the router does not drop UDP states faster than 30 minutes, and also assuming that the provider is sending you invites on the registered port rather than always on 5060, there should not be a need for an inbound port forward to Asterisk, and you should not need insecure=port.
The invite option disables authentication - which means only that Asterisk will not force a check of the password on the other end. Where the IP address is well known and trusted, the extra overhead and delay of authenticating incoming INVITEs is not needed. On Thu, Apr 2, 2015 at 2:28 PM, Daniel Heckl <[email protected]> wrote: > Scott, I have changed the configuration as said it and will test it. I’m > curious. > > Can you briefly explain what insecure=invite,port does? > > ;insecure=port ; Allow matching of peer by IP address without > ; matching port number > ;insecure=invite ; Do not require authentication of incoming INVITEs > ;insecure=port,invite ; (both) > > Do I understand correctly that in this mode the IP address is not checked > and no authentication is required? > > Am 02.04.2015 um 20:11 schrieb Scott Griepentrog <[email protected] > >: > > I'd be curious if setting > > insecure=invite,port > > makes any difference either (without alllowguest on). > > > On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl <[email protected]> > wrote: > >> Ok, I have tested dnsmgr. This is not a solution, the situation has not >> changed. With dnsmgr I can not place outbound calls. I do not know why and >> what dnsmgr really do. >> >> My current solution is as follows: >> >> Say allowguest=yes, configure the default context that there can not be >> placed outbound calls. Use iptables to DROP all at your SIP port and allow >> only your local phones and the sip trunk ip range. I think srvlookup must >> be set to yes to place outbound calls if there is an ip address change. >> >> I think with the restriction of the firewall that should be a secure >> solution. >> >> > Am 01.04.2015 um 19:23 schrieb Sebastian Kemper <[email protected]>: >> > >> > On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: >> >> On 4/1/15 10:48 AM, Daniel Heckl wrote: >> >>> John, >> >>> >> >>> thank you four your answer. I think you have misunderstood the >> >>> problem. It’s about a ip address change of the sip trunk, not of my >> >>> asterisk server. >> >> You would probably benefit by enabling the DNS Manager to allow for >> >> dynamic IP changes: >> >> >> >> # cat dnsmgr.conf [general] enable=yes ; enable creation >> >> of managed DNS lookups ; default is 'no' refreshinterval=180 ; >> >> refresh managed DNS lookups every <n> seconds ; default is 300 (5 >> >> minutes) >> > >> > Hello Andres, >> > >> > I read that same suggestion elsewhere in connection with Deutsche >> > Telekom, so it seems there's some benefit in it. >> > >> > Daniel, did you try it out already? >> > >> > Kind regards, >> > Sebastian >> > >> > -- >> > _____________________________________________________________________ >> > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> > New to Asterisk? Join us for a live introductory webinar every Thurs: >> > http://www.asterisk.org/hello >> > >> > asterisk-users mailing list >> > To UNSUBSCRIBE or update options visit: >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > [image: Digium logo] > Scott Griepentrog > Digium, Inc · Software Developer > 445 Jan Davis Drive NW · Huntsville, AL 35806 · US > direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 > Check us out at: http://digium.com · http://asterisk.org > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- [image: Digium logo] Scott Griepentrog Digium, Inc · Software Developer 445 Jan Davis Drive NW · Huntsville, AL 35806 · US direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090 Check us out at: http://digium.com · http://asterisk.org
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