Hello all , can somebody provide screenshots from Telekom SIPsettings, thanks.
What are Telekom SIP settings? Do you mean proper settings for Deutsche Telekom?
That said and in case you do mean DTAG, they have different VoIP products with different requirements as far as the configurations go.
Yes i mean Telekom , this Product calls “MagentaZuhause XL”
Please post the configuration information you got from DTAG, including their proxy. “Zuhause” does not sound like one of their products for companies, so your proxy might be tel.t-online.de. If it’s not tel.t-online.de also post the result of dig tel.t-online.de ANY .
I actually meant dig proxy ANY.
I’d recommend to start with SIP over TCP/IP and to get a real proxy using _sip._tcp.tel.t-online.de. You will get 3 servers like d-epp-100.edns.t-ipnet.de and this is what you should use as your proxy in the configuration. Other than that you should follow the Telekom configuration notes and configure normal inbound and outbound routes. You do not need to evaluate the P-Called-Party-ID header in this case as is necessary for German Telekom “trunk” services. It doesn’t exist for this service.
The background is as follows. There are actually more than the three SIP servers in the t-ipnet.de net and for subsequent DNS calls they rotate and that includes also the priority. Since the underlying Asterisk core always evaluates the addresses you will end up talking to a SIP proxy where you have not registered to with a certain probability. DTAG does not synchronize the registration credentials across several proxies. To solve this problem you need to patch the Asterisk source code a bit such that only the IP of the associated registration gets used for any subsequent communication. This implies that for a short period of time you can be registered to 2 or (less likely) more Telekom servers.
With this patch you could setup FreePBX such that Asterisk automaticall selects the recommended transport, TLS included. So far I haven’t seen anybody having done that.