8 thoughts on - Change By Deutsche Telekom End Of Februar. Can Someone Help Me?

• Hi!

  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

  —

• Am 15.02.2021 um 21:40 schrieb Michael Maier:

  Hi Michael,

  Mmm… I’m using tel.t-online.de, but I’m not sure I’m using pjsip…

  module show say me:

  res_pjsip.so Basic SIP resource
  46 Running core

  Do I use pjsip?

  You mean, I have to create a “fake” Zone tel.t-online.de in my Bind with these settings? Looks like dangerous, if they changes something…

  Thanks Luca Bertoncello
  (lucabert@lucabert.de)

  —

• Am 16.02.2021 um 19:56 schrieb Michael Maier:

  Hi Michael,

  gw*CLI> pjsip show registrations No objects found.

  So I don’t use pjsip… 🙁
  Maybe could you send me an abstract of your configuration?

  So if I undestand what you mean, you check the NAPTR and SRV für
  _sips._tcp.tel.t-online.de and save the record in a “virtual domain”
  rpz-tonline, is it correct?
  Then I suppose you use this domain instead of tel.t-online.de in the SIP
  configuratione as “host”, “outboundproxy” and “fromdomain”, is it correct?

  OK, I’ll check it…

  Could you explain me that? I’m not an expert of Asterisk… 🙁

  Thanks a lot!
  Luca Bertoncello
  (lucabert@lucabert.de)

  —

• Yes.

  Take a look here [1]

  No – you have to use the correct domain name in asterisk. Only bind knows about the fake domain. You have to configure bind correctly.

  You have to create the fake domain in the bind config like this:
  options {
  …
  response-policy {
  zone “rpz-tonline”;
  };
  };

  …

  zone “rpz-tonline” {
  type master;
  file “/var/named/rpz-tonline-override”;
  allow-query { any; };
  allow-transfer { any; };
  allow-update { any; };
  };

  All other things: take a look at the script! It’s not that complicated.

  Well, if you want to use it, you really should engage yourself a bit more to get it solved. It’s not that easy. Or you may forget about the DNS fake and live with the problem, that asterisk could partly switch sometimes to another server –
  breaking the telephony. I don’t think it would happen that often, because Telekom usually is extremely stable. Try at first to get a running pjsip configuration. The DNS theme could be done later on.

  Regards Michael

  [1]
  https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234

  —

• Am 16.02.2021 um 22:32 schrieb Michael Maier:

  Hi Michael

  So, maybe I got it… I tested the configuration with my Fax number and it seems to work (= I
  can call the fax and can call my mobile phone from the fax with
  “originate…”).

  On the registration I have:

  [pbxfax]
  type = registration retry_interval = 20
  max_retries = 10
  contact_user = 00493514977291
  expiration = 120
  transport = transport-udp outbound_auth = pbxfax client_uri = sip:03514977291@tel.t-online.de server_uri = sip:tel.t-online.de

  First: can I use tel.t-online.de or _MUST_ I change it? If I understand your previous E-Mail, I’d say that I can leave tel.t-online.de…

  Then I have a question by the Dialplan… Currently I have:

  [fax-out]
  exten => _X.,1,NoOp()
  exten => _X.,n,Verbose(2,Call from FAX)
  exten => _X.,n,Dial(SIP/pbxfax/${EXTEN},,R)

  And I’ll replace it with:

  [fax-out]
  exten => _X.,1,NoOp()
  exten => _X.,n,Verbose(2,Call from FAX)
  exten => _X.,n,Dial(PJSIP/pbxfax/sip:${EXTEN}@tel.t-online.de,,R)

  Is it correct? I tried with
  “PJSIP/pbxfax/pjsip:${EXTEN}@tel.t-online.de,,R” and it does NOT work… Is it correct, that I have to leave “sip:…”?

  Thank you very much for your help!!
  Luca Bertoncello
  (lucabert@lucabert.de)

  —

• Congrats!

  No, you mustn’t change it. You must use tel.t-online.de.

  Correctly!

  Don’t know – I don’t care about dialplan – I’m using FreePBX 🙂

  Thanks Michael

  —

• Am 18.02.2021 um 18:59 schrieb Michael Maier:

  So, it seems it does NOT work as expected… I tried to activate the FAX and it works, then I activated my number and it works, too. Finally I activated the number of my wife and it does not work anymore… If I call the number I can only see (verbose 42):

  [Feb 18 19:57:12] NOTICE[19379] res_pjsip/pjsip_distributor.c: Request
  ‘INVITE’ from ‘‘
  failed for ‘217.0.21.64:5060’ (callid: p65550t1613674632m753568c93349s2)
  – No matching endpoint found

  and no phone rings… After that, even if I restore the single number to SIP I only get the error and nothing work, until I restored _ALL_ numbers to SIP.

  Do someone has an explanation and (better!) a solution to the problem?

  Thanks Luca Bertoncello
  (lucabert@lucabert.de)

  —

• You have to do all of the configuration mentioned here[1] for *each* number. Afterwards, you have to route the incoming call to an internal device. As I’m using FreePBX, I don’t know how to do it *correctly*.

  Solution:
  You have the choice between: programming your PBX yourself (and have the struggle and pain) or let this pretty difficile job do others for you – they provide extremely good solutions for a lot of telephony features – it makes no sense to reinvent those features without having the required knowledge – so, use FreePBX. But it’s of course your decision.

  [1]
  https://www.ip-phone-forum.de/threads/hilfe-f%C3%BCr-grundeinstellung-asterisk-telekom-ben%C3%B6tigt.307115/post-2374234

  —