FreePBX 13.0.192.19 + Asterisk 13.22.0 - German Telekom (Deutsche Telekom) - no outgoing calls?

FreePBX Tips and Tricks
1

Hi,

FreePBX 13.0.192.19 + Asterisk 13.22.0
Registration of Telekom numbers: works OK, shows “Registered” (PJSIP)

Calling out always ends with 403-Forbidden - chan_sip AND pjsip.

There must be something wrong somewhere deep in the configuration.

all variations of username/password (there are several possibilities reported in various blogs…) do not work.

Asterisk is running behind a firerwall and NAT is needed. The external IP can be determined using the dyndns name of the external interface

I really don’t care whether I call out using chan_sip or pjsip - but corrently neither of them works.

Any ideas? There must be a solution, I’m sure …

Regards

Chris

2

Show us the failure in the logs. That will help.

3

well, magic restart helped… scratch the whole box (there was an harddisk error), set up new - works.

no idea why

4

Hi @bofh42,

From my experience with Deutsche Telekom (DT), here are my notes for a working SIP trunk (not PJSIP) configuration:

there is a DNS resolving problem (add the record to the hosts file):

217.0.15.67 sip-trunk.telekom.de

*****

Duetche Telekom Supports TCP only and user phone:

host=sip-trunk.telekom.de
defaultuser=551234567890
secret=xxxxxxx
type=peer
port=5060
fromdomain=sip-trunk.telekom.de
fromuser=+4981234567890
canreinvite=no
insecure=port,invite
disallow=all
allow=alaw&ulaw
session-timers=refuse
qualify=yes
transport=tcp

*****
register with tcp and phone as user:

tcp://+4981234567890:xxxxxxx:[email protected]/+4981234567890

****

We need to activate dns srv lookup in sip.conf

Thank you,

Daniel Friedman
Trixton LTD.

1 Like
5

Thanks, but …

I’m not using [email protected] but a “normal” tel.t-online.de … which is a bit different, it seems

6

It really looks like it does not matter what you put into the authentication fields - as long as you are connected from your own DSL/VDSL line.

All variants work at the moemnt.

The “trick” was to complety reboot the asterisk machne as soon as connection problems come up … :frowning:

8

Hi @bofh42,

The t-online is a white label of Duetche Telekom and as such my configuration is correct. Anyhow, here is another source for interconnecting to DT with Asterisk: https://www.voip-info.org/asterisk-settings-for-t-onlinede/

Thank you,

Daniel Friedman
Trixton LTD

9

sip-trunk is used for telekom sip trunk agreements. (Basic number with extensions, earlier called ISDN Anlagenanschluss) and can be used from any dsl/vdsl location). tel.t-online is used for single connections, single numbers (Telekokm all-ip, earlier MSN, Mehrgeräteanschluss) and works from the DSL/VDSL connect where it’s contracted to --only (!), with no real need to send legitimation. So everything works.