Call problem outing with italian provider Windtre. Timeout

FreePBX Providers
1

Salve, ho un rasperry con asterisk 22 e freepbx 17.

Gli errori

WARNING[1037]: res_pjsip_outbound_registration.c:1570 handle_registration_response: 403 Forbidden fatal response received from ‘sip:voip.windtre.it:5060’ on registration attempt to 'sip: …

sembra essere un problema windtre, le chiamate parte ma mi da timeout: SIP/2.0 504 Server Time-out

Ho provato a impostare il realm su voip.windtre.it ma niente. qualcuno è riuscito?

quando chiamo freepbx mi da un messaggio dicendo che tutti i circuiti sono occupati ma la verità è che mi va in timeout la risposta da parte di windtre.

Chiamare l’assistenza windtre è inutile tanto non ci capiscono niente e prendono in giro dicendo che chiameranno per risolvere. fanno perdere solo tempo. l’unica cosa da fare è cambiare gestore.

I trunk sono correttamente registrati e per le chiamate in entrata è tutto apposto.

Qualcuno è riuscito a configurare correttamente le chiamate in entrata ? le configurazioni sono corrette user pass parametri etc.

2

Salve,
ho rasperry con asterisk 22 e freepbx 17.
Sono registrato sul trunk windtre e ricevo le chiamate senza problemi.
Il problema è nelle chiamate in uscita.
errore:
WARNING[1037]: res_pjsip_outbound_registration.c:1570 handle_registration_response: 403 Forbidden fatal response received from ‘sip:voip.windtre.it:5060’ on registration attempt to 'sip: …

a quanto pare è un problema di windtre. asterisk manda l’invito ma windtre fa andare in timeout

SIP/2.0 504 Server Time-out - S - 16046

inutile l’assistenza clienti sono tutti incompetenti anzi… fanno perdere solo tempo dando informazioni errate. l’unica soluzione sarebbe passare ad un altro operatore..

qualcuno è riuscito configurare le chiamate in uscita con le ultime versioni di asterisk/freepbx ?
pare che il problema sia proprio di windtre…

3

Hi, I have a rasperry with asterisk 22 and freepbx 17.

Errors

Warning [1037]: res_pjsip_outbound_registration.c: 1570 Handle_registration_RESPONSE: 403 Forbidden Fatal Response Received from ‘SIP: VoIP.Windre.it: 5060 ’ON REGISTRATION INTERMPT TO’ SIP: …

It seems to be a Windtre problem, the calls part but give me Timeout: SIP/2.0 504 Time-Out Server

I tried to set the realm on voip.windre.it but nothing. Has anyone succeeded?

When I call freepbx it gives me a message telling that all the circuits are busy but the truth is that I am in timeout the response by Windtre.

Calling Windtre assistance is useless so much they don’t understand anything and make fun of saying that they will call to solve. they only waste time. The only thing to do is change provider.

The trunks are correctly recorded and for incoming calls everything is affixed.

Has anyone managed to correctly configure incoming calls? The configurations are correct user pass parameters etc.

4

Howdy! Welcome to the forums.

Please consider posting more SIP logs – but scrubbed of personal information – via our pastebin.freepbx.org site.

You may also want to double-check your user/pass for special characters that could be preventing registration/authentication, along with your firewall for any dropped packets, and maybe another test trunk with a different telephone company provider.

5

sorry. the log is here: https://pastebin.freepbx.org/view/f163846b

6

You are trying to call a zero length phone number.

This is the result of misconfiguring the outbound proxy.

Firstly, is there actually an outbound proxy at all?

If there is, chan_pjsip requires the outbound proxy to be specified as it would appear in a Record Route header. These days that would always be with loose routing, so ;lr needs to be added. Some providers don’t like seeing Route headers, even with loose routing syntax, in which case you also need to add ;hide.

As this parameter gets placed into an Asterisk .conf file, and ; starts a comment, the ; needs to be escaped with a back slash. Where it is actually required (i.e. the Server setting doesn’t resolve to the first hop machine address), it would be typically in the form:

sip:sip.provider.example\;lr\;hide

<--- Transmitting SIP request (1033 bytes) to UDP:151.6.119.246:5060 --->
INVITE sip:voip.windtre.it SIP/2.0
.....
Route: <sip:[email protected]:5060>
2 Likes
7

Yes! But please fix two settings in the pastebin:

  1. Enable “Never” as an option for Delete After.
  2. Set “Never” as the default expiration.

pastebin.freepbx.org used to work like this, but when it went down and got moved to another server, someone forgot to preserve these settings.

IMO, it is essential for logs or other pastings to be available for future readers, so they can follow along. So, when it got broken, I started to recommend pastebin.com as an alternative. Unfortunately, their onerous tracking policy is causing several expert members to not read those posts, so please fix the Sangoma pastebin.

9

To clarify, what was needed was a backslash: \

Agree (mostly) on 1 but not 2 :balance_scale: because Never is a long time.

12

That was exactly the problem. The backslash \ before the ; .It had to be inserted in pjsip trunk > pjsip setting > advanced > outbound proxy: sip:voip.windtre.it\ ;lr\ ;hide .(without space.. this forum :poop:)
Before it was sip:voip.windtre.it;lr;hide. It was driving me crazy.
Now I just have to configure the dial patterns which are now on . (it takes everything). Thanks very much :hugs:

PS: The maximum reading time is one month. You cannot enable “never” right now..

Sorry for duplicated posts, but this forum does not allow to make any changes.. :face_vomiting:
i wrote \ ;lr \ ; hide …what a f*ck

13

Try this:

sip:voip.windtre.it\;lr\;hide

For specific code formatting of strings, please consider the </> button and/or directly typing backticks: `

…which will render text like this \; ooh neat! \; and if you use three of them on a line by itself

then you can make a
whole block of monospaced text
until you close it

…with three more backticks ``` on a line all by themselves.

New users are limited in different ways to help prevent spam and for other purposes.