Scott, thank you four your reply. 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.
Only after a sip reload the peer works again. That can't be normal... Daniel > Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <sgriepent...@digium.com>: > > You have two options for dealing with an IP change during the registration > period: > > 1) set the registration time to shorter period of time to minimize the > downtime > > 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. > >> On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl <daniel.he...@gmail.com> wrote: >> Maybe someone could elaborate on my first question again. >> >> 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? >> >>> Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.he...@gmail.com>: >>> >>> Hello Sebastian, >>> >>> 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. >>> >>> 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. >>> >>> If I change insecure to insecure=port,invite - could that be a solution? >>> >>> 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? >>> >>> Daniel >>> >>>> Am 30.03.2015 um 20:09 schrieb Sebastian Kemper <sebastian...@gmx.net>: >>>> >>>> On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: >>>>> Hello >>>>> >>>>> I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom >>>>> Germany. We have sometimes problems with incoming and outgoing calls. >>>>> I hope I can explain it understandable. >>>> >>>> Hello Daniel, >>>> >>>> I'll find myself in the same situation a few weeks from now :-) >>>> >>>>> >>>>> For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de >>>>> <http://tel.t-online.de/>), the message is answered with OK and the >>>>> peer is registered. >>>>> >>>>> Usually INVITES comes now from this ip address. All works fine. But >>>>> sometimes INVITES comes from an other IP address, for example >>>>> 217.0.23.100. This request Asterisk responds with 401 Unauthorized. >>>>> >>>>> In the next register procedure REGISTER are sent to the new ip address >>>>> and answered also with OK. But qualify OPTIONS are continue be sent to >>>>> the old ip address. Incoming and outgoing calls are canceled. Outgoing >>>>> calls are answered with Forbidden. >>>>> >>>>> Even if the REGISTER procedure works with the new ip address, the >>>>> peers are connected with the old address. >>>>> >>>>> Waiting doesn’t help, only a „sip reload“ update the ip address of the >>>>> peer. >>>>> >>>>> What is the solution for this problem? How can asterisk update the >>>>> peer? >>>> >>>> I think the solution - for the inbound issue at least - could be to add >>>> more hosts as a peer. Have a looks at this forum post: >>>> >>>> http://www.ip-phone-forum.de/showthread.php?t=268787&p=1999371&viewfull=1#post1999371 >>>> >>>> The user used a template and than he added peers, each with its own IP >>>> address. The provided list was last updated in 2014, though, so I assume >>>> the provider in the meantime has added to that list. >>>> >>>> It looks pretty tedious, though, I mean there could be dozens of IPs >>>> you'd have to add. But I guess this is the way to go with Asterisk 11 >>>> and chan_sip. >>>> >>>> The future looks brighter :-) I read that with pjsip, which I understand >>>> is the replacement for chan_sip, you can have one peer entry and match >>>> an IP range instead of a single host. That should tidy up the dialplan. >>>> >>>> What I'm a little afraid of is the SIP provider using IPs out of a range >>>> that they also use for other services. Maybe out of the same range they >>>> hand out IPs to their customers. I guess we got to be careful :-) >>>> >>>> Kind regards, >>>> Sebastian >>>> >>>>> The Asterisk is local behind a NAT with a firewall, following settings >>>>> are used: >>>>> >>>>> externhost with DynDNS stun with stun.t-online.de >>>>> <http://stun.t-online.de/> nat=yes srvlookup=yes allowguest=no >>>>> trustrpid=no insecure=invite qualify=yes >>>>> >>>>> Thank you! Daniel >>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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 >> >> -- >> _____________________________________________________________________ >> -- 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 > > > > -- > > 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
