[asterisk-users] Change by Deutsche Telekom end of februar. Can someone help me?

[Michael Maier]

Hi!

On 15.02.21 at 08:43 Luca Bertoncello wrote:
> Hi list!
> 
> I received a letter from Deutsche Telekom where they say me, that I need
> to change "something" on my router until 28.02.2021, otherwise I cannot
> phone anymore.
> Since I use Asterisk and I don't have a router, I'm not sure what I need
> to do...
> In the letter there is an URL to "explain" how to change the
> configuration if I use a VoIP-phone, but they only say, that I don't
> have to use Port 5060, but Port 0...
> 
> Surely there are in this list someone other using Deutsche Telekom...
> Does someone of them understand what I should change in the Asterisk
> configuration?

They're switching to DNS NAPTR / SRV[1]. If you are using Asterisk /
pjsip and hostnames (tel.t-online.de e.g. for the AllIP service), you
won't have any problem (using asterisk 14 or higher), because it's
default. But you may have problems with the handling of the calls,
because Telekom needs the client always to use the same server for all
activities after the register has been done (the SRV entries contain 3
servers and asterisk will use them "randomly" if it detects a problem -
regardless which server of the list has been used for registration -
this won't work with Telekom and will lead to not working outbound calls
/ interrupted calls e.g.). This won't happen very often (because they
have been extremely stable in the past), but I could see it nevertheless
already. If you want to be really sure to not face this problem, you
have to create a workaround by adding a rpz zone e.g. with an own bind,
which is fed by an own job and presents asterisk just one server when
looking up the SRV entries after the NAPTR call. NAPTR / SRV works like
this (example for tel.t-online.de):

1. Search for the service names
dig noall +answer tel.t-online.de NAPTR
tel.t-online.de.        5       IN      NAPTR   10 0 "s" "SIPS+D2T" ""
_sips._tcp.tel.t-online.de.
tel.t-online.de.        5       IN      NAPTR   30 0 "s" "SIP+D2T" ""
_sip._tcp.tel.t-online.de.

2. Take the answer of the NAPTR output (TCP/TLS, TCP)
dig +noall +answer _sips._tcp.tel.t-online.de SRV
_sips._tcp.tel.t-online.de. 2234 IN     SRV     10 0 5061
s-eps-110.edns.t-ipnet.de.
_sips._tcp.tel.t-online.de. 2234 IN     SRV     20 0 5061
h2-eps-100.edns.t-ipnet.de.
_sips._tcp.tel.t-online.de. 2234 IN     SRV     30 0 5061
d-eps-100.edns.t-ipnet.de.

dig +noall +answer _sip._tcp.tel.t-online.de SRV
_sip._tcp.tel.t-online.de. 3600 IN      SRV     30 0 5060
d-epp-100.edns.t-ipnet.de.
_sip._tcp.tel.t-online.de. 3600 IN      SRV     10 0 5060
s-epp-110.edns.t-ipnet.de.
_sip._tcp.tel.t-online.de. 3600 IN      SRV     20 0 5060
h2-epp-100.edns.t-ipnet.de.

Asterisk now must use always the same server for all activities to
Telekom - like register, invite, options - but that's not yet supported
by Asterisk - therefore you have to ensure, that asterisk always uses
the same server. Easiest way is to provide just one in the DNS answer ... .


Regards
Michael

[1]
https://geschaeftskunden.telekom.de/hilfe-und-service/online-services/hilfe-internetanschluss/telefonieanpassung#telekom