Hi Kai,

I am not a fan of such integrated solutions. Before we used Freeswitch we had an all in one PBX from Alcatel with integrated Internet Router/Firewall. The Service was crap and when we wanted to do something special it wasn’t working or costed a lot. We use Openbsd as router/firewall now. The entire VOIP runs in a separate VLAN. Have you successfully set up freeswitch without using the Digibox?

Hi Paul,

got it running in several sites without major problems. As you said transport is either TCP or TLS. Faxing is basically passthrough only as the do not offer gateway services and I did not find a subscriber terminating T.38. If you use their provided equipment (Digibox) you so not need to setup anything special as the port mapping will be handled by the box itself (SIP and RTP gateway included).

Hi Paul,

Good to here that Keep-Alive is working. I found some reports on the internet where people opened ports from the firewall and wanted to restrict the IP range. However, incoming and outgoing servers seem to change frequently. All in all, it is probably not possible to narrow this down to 100%, which is why this is not good "protection".

You may find out additional information about the trunk when querying DNS for SRV and NAPTR records.


Hi Giovanni,

I haven't dealt with kamailio yet. The possibilities seem to be almost limitless, as the functions go far beyond a simple proxy.
But setting up a simple proxy shouldn't be so difficult ? I have already invested a lot more work in the setup of Freeswitch than
a normal plug and play user would do. Unfortunately, I can't find much information about this topic in connection with Freeswitch.

There are not ready made secure way to implement sip, particularly and especially a sip proxy. 



2018-01-28 18:40 GMT+01:00 Kai Danielmeier <kai.danielmeier@gmail.com>:
Hi Paul,

got it running in several sites without major problems. As you said transport is either TCP or TLS. Faxing is basically passthrough only as the do not offer gateway services and I did not find a subscriber terminating T.38. If you use their provided equipment (Digibox) you so not need to setup anything special as the port mapping will be handled by the box itself (SIP and RTP gateway included). 

BR

Kai

2018-01-27 18:55 GMT+01:00 Paul Muaddib <paul.muaddib83@gmail.com>:
Hi,

does anyone here have experience with the SIP trunk service from the German Telekom: "DeutschlandLAN SIP-Trunk"? There is little information from Deutsche Telekom on the Internet besides the "Technical Technical Specification of the SIP - 1TR114" As I have read so far, Deutsche Telekom has not necessarily been sticking to the standard.

What I know so far

Transport protocol is probably TCP instead of UDP
Codecs: G. 711a and G. 722
Fax either via G. 711a or T. 38

My Setup
Internet <-----> Router/Firewall(NAT) <----> Freeswitch

According to the instructions, they require you to open several ports in the firewall. Is that really necessary? Does the Keep-Alive function work without opening any port?

At the Federal Office for Information Security (BSI) it is additionally recommended to use a SIP proxy such as kamailio

At the moment we are still using an ISDN gateway. I don't like to run into problems when we switch over, because I can't go back then.

Best regards,
Paul

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