[asterisk-users] Update peer IP address

[Daniel Heckl]

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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> 
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