VDSL-Modem im Bridge-Modus: Anschluss mit Telefonie

[Deafty]

Thanks heaps.

Will do and don' worry I will be polite

 

[Deafty]

Okey Dokey:

Fritz box up and running. including telephone (DSl direct to box)
Now configured as IP client.

with UpnP enabled on asus (not liking this )
Telephone can make calls.
BUT Telephone can not receive calls

In the Fritz GUI I can see a call coming in.

I was expecting to have to configure the port but I have no idea what port vodafone uses?

Any more ideas?

 

[norKoeri]

In the Asus, you have to disable the ALG ‘SIP’ and in the FRITZ!Box you have to keep the port alive: fritz.box → Telefonie → Eigene Rufnummern → Anschlusseinstellungen → (Telefonieverbindung) Einstellungen ändern → Portweiterleitung des Internet-Routers für Telefonie aktiv halten: 30 Sekunden. Did you do that already?

I was expecting to have to configure the port but I have no idea what port vodafone uses?

No, not that way, please.

 

[Deafty]

Well before I read your post. I did this.

• enabled UpnP on asus router
• then as you mentioned,,. In Fritzbox enabled the portforwarding to be active (for 30 seconds)

Then I could receive calls. but landline had no audio. (ALG Sip is dissabled.)
currently looking into it.....

 

[Deafty]

I'm not hearing incoming audio, It is perhaps a port issue.

Why may I ask are you insisting on me not making manual port changes?

Im at a loss here,....

Edit: I temporaily placed the fritz box in a DMZ and issue persisted.
It is now dissconnected and I will reset the device to factory defaults

Edit2: from reading it seems that this is common when a blockage of RTP ports occurs

Edit3: Most likely cause is I'm still double nated in my everyday setup. So will connect with single NAT. this should fix it. at least fingers crossed

Edit 4:

Yay that fixed it.. rofl thanks for all your help and pointing me in the right direction

++++++++
I'll be moving on to my Omada setup which arrived last week and have been configuring offline.
++++++++

 

[Deafty]

So here is my new (uninstalled) hardware.

Omada:

• Router:
  • ER707 M2
• L3 Switch:
  
  • SG3428 (24 port)
• L2 PoE Switch :
  
  • SG2210MP
• WAP:
  
  • EA615
• Contoller:
  • OC200

It did take me quite a while to ensure that I obtained all the correct hardware versions.
(Hardware versions are now printed on the box so you can send it back if you receive the wrong one)

In particular the SG3428 Switch.
It is currently still sold here under the TL-SG3428 label which you may notice is listed as EOL.
As long as You obtain Hardware version 2.2 or higher you can upgrade the firmware (from Global site!) and the switch will be rebranded to the Omada SG3428 with HW version 2.3

Anyway I am currently setting up my configs then will have to repatch my 10" rack into a new19" one
Run a few more cables for security cams and find a way to patch my horrible cat5 intercom cable that the builders made a mess of.

Entire house had Cat7 (was cheaper than Cat6 or 6a single cable) duplex run into every room by myself when we built a few years back. I've done enough work with Telecon cable runs for this to be straightforward enough.

Anyway onwards and upwards....

@ norKoeri: Thanks for all your help with VOIP my knowledge in this area was non existent.

 

[norKoeri]

Why may I ask are you insisting on me not making manual port changes?

Just to answer that one: VoIP audio packets float via RTP.

Nowadays, RTP ports are created dynamically. Therefore, you cannot determine the actual port number beforehand. Therefore, everyone—at least all VoIP clients which have not ignored that last 30 years—use UDP-hole punching. There are exceptions to that rule, yes. Especially as some device still use static RTP ports. Nevertheless because you do not know whether the firewall does a port remapping itself, this is useless knowledge. To summarize, you should always start with no port forwarding, no exposed host, no DMZ. When that fails, please, do not try-around but rather debug why the state-of-the-art does not work. If that is broken, you miss incoming audio within a call, the RTP is blocked by a firewall.

Similar applies to SIP, the VoIP controller packets. Although you connect to port 5060 normally, the originating port could be dynamic as well, a firewall might remap. On top of that connection, the VoIP client uses keep-alives, so no firewall closes the connection. Again, that is the state-of-the-art. If that is broken, you miss incoming calls, the incoming SIP is blocked by a firewall.

All those guidelines about ports and port forwarding are factoid. Their authors are either liars, have not updated their knowledge for 25 years, or have not updated their webpage for 25 years. And yes, there are exceptions as some VoIP software is still not using that. Nevertheless, AVM FRITZ!OS is not one of them.