FreePBX on Sunrise Switzerland

Hello,

I live in Switzerland and just went to change my Internet provider from Swisscom to Sunrise. For the telephony with the old one I had a FreePBX running on a Raspberry, SIP credentials and everything worked fine.

For the new provider I did setup a new FreePBX, running virtualized on my QNAP. All the extensions now are migrated to the new installation and are working internal, configured as PJSIP.

I’m experiencing some difficulties in configuring the SIP Trunk. I have the following SIP credential from the provider:

Telephone number: 091xxxxxxxx

Username: V0yyyyyyyyy

Domain: gra.ims.sunrise.ch

Proxy: gra.ott.sunrise.ch

Secret: zzzzzzz

After some trials I suspected that my account could be blocked, but I configured a Gigaset Go-Box 100 (for 2 DECT phones) to direct with the credentials and it works.

In the Log I have the following error:
res_pjsip_outbound_registration.c: 403 Forbidden fatal response received from ‘sip:gra.ims.sunrise.ch:5060’ on registration attempt to ‘sip:[email protected]:5060’,

Does somebody have any suggestion?

For the configuration for the trunk I have chosen SIP(chan_pjsip) with the following configuration:

PJSIP:

Advanced:

Codecs:

In the Asterisk SIP settings, I have the following config:

Thank you
Paolo

I know nothing about Sunrise, but the Username and Auth username may be backwards; Username is typically a phone number and Auth username is usually an account number.
In which fields did you enter these values for the Gigaset?

Otherwise, perhaps a routing issue with Outbound Proxy. Try setting it to
sip:gra.ott.sunrise.ch\;lr\;hide
(note backslash semicolon in two places).
If you still have trouble, post a SIP trace showing the registration sequence from the Gigaset.

Hi @Paolodic,

A few things that come to mind, have you disabled your PBX Firewall to see if that makes any difference, I feel like it may also have something to do with you Auth Username and Username.

Hi Stewart,
thank you for quick answer. Maybe in one of many trials I did, there was the version you propose, but at a certain point I thouht to be banned…
Ok, I will try this way. I’ll let you know.

The working Gigaset config is:

It looks pretty simple:

  1. Shut down the Gigaset connection, so you don’t have a conflict.
  2. Set Username to the v014 value.
  3. Leave Auth username blank (will use the same value as Username).
  4. Make sure that Secret matches what you have for Authentication password on the Gigaset.
  5. Set Outbound Proxy as noted in my previous post.
  6. Set Expiration to 180.

With luck, it should now register. If not, post the Asterisk log for a registration attempt, with pjsip logger turned on.

If it registers but you can’t make calls, also try:

  1. Set From User to the same value as you have in Username.
  2. Set From Domain to the same value as you have in SIP Server.

Hi Stewart,
one picture is better than 1000 words:

Thank you very much!

Now I will setup the in-outbound Routes.

Just one question: what does it mean “\;lr\;hide” in the outbound proxy?

BR
Paolo

As you suspected, I had to do “From user” and “From domain” for making calls outside.
For receiving calls I had to set “Allow Anonymous Inbound SIP Calls” and “Allow SIP Guests” in the general SIP Settings (Asterisk Settings).

So it seams now that everything works fine. I will do further checks in the next days.

Also thank to @Taylor: I already had disabled the firewall. Now is time to try to reactivate it and see what happens!

Just for the records, I’m running:
FreePBX 15.0.37.9
Asterisk 16.13.0

Thank you again
Paolo

This is a significant security vulnerability; you should track down why the incoming call is not being associated with your trunk and fix that.
You might try setting Match (Permit) to the list of addresses from which Sunrise can send calls.
If the calls are coming from 195.141.201.129, I don’t understand why they aren’t being recognized.

Hi Stewart,
Thanks for your suggestion. I have set Match with the address from Sunrise and it still works! So the Anonymous Inbound SIP callss is no more active.

BR
Paolo