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 <[email protected]>: > > 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
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