[asterisk-users] Update peer IP address

Scott Griepentrog sgriepentrog at digium.com
Tue Mar 31 15:45:45 CDT 2015


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.heckl at 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.heckl at 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_ml at 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
>
>
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Scott Griepentrog
Digium, Inc · Software Developer
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