SIP status 403 forbidden

Hi guys, I was using Freepbx with SIP ITSP. When using SIP trunk, incoming calls works. But when making outgoing calls, it says “all circuits are busy now.” After analyzing with wireshark, it shows SIP status 403 forbidden. Please suggest me how can I make outgoing calls using SIP trunk.

Hi @kyaw
I think you are missing some Outbound Route CID and Dialpatterns settings. And are you able to add here Asterisk dial log.

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This can represent many different types of problem. The full text of the message, preferably taken from the Asterisk full log, with “pjsip set logger on” (or equivalent for deprecate drivers) in effect, may give more information. However possibilities are:

  1. ITSP identifies by IP address and you are not sending from the address you arranged with them.
  2. As above, but the Via address represents an internal address, because you have incorrect NAT settings.
  3. As first one and your caller ID is not approved.
  4. As first one and your outgoing number format is invalid.
  5. As first one and ITSP doesn’t like your Contact header.
  6. ITSP identifies by user name and the user name is invalid (unlikely as this would cause registrations to fail, and you have either registered, or you are being identified by IP address.
  7. You have no credit for outgoing calls
  8. ITSP doesn’t like you for other reasons.,
  9. etc
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Here is the interesting part. I changed from FreePBX to Yeastar P-series software edition PBX, all the incoming and outgoing calls are working perfectly. But I don’t want to use Yeastar P-series software edition due to license pricing.

Try setting From User to the same value you have in Username and From Domain to the same value you have in SIP Server.

For chan_sip (not recommended), the parameters are fromuser and fromdomain.

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The call to 09977270397 (AFAICT, a valid Aksu number), was made from extension 1001 running Eyebeam.

When the INVITE was sent on the trunk, the From header was
From: sip:[email protected];tag=b33bb3dc-fb53-48b2-8d94-6e6c19526f59
and the provider responded with
SIP/2.0 403 Forbidden

Warning: 399 “The caller IMPU is invaild,And caller replacement policy in trunk is rejection”

IMPU (IMS Public User Identity) is usually a phone number, but depending on the provider, it may be an account number or similar identifier. 1001 is definitely incorrect.

I had previously suggested

but you did not acknowledge it. What happened when you tried, or why were you unable to try?

If you can’t easily resolve this, please post a log (with pjsip logger on) of an incoming call. The headers a provider sends on incoming usually show what they will accept on outgoing.

BTW, the useful log information was in the post you (accidentally?) deleted – in the most recent post, the call had already failed.

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Sorry @Stewart1 , I don’t know where can I edit that information. Please enlighten me.
P.S SIP ITSP provide only IP Address for SIP Trunking.

What is your Outbound Routes, could you pls past here screenshot just Route settings page.

what is that number ? and are you able to test with Override Extension: YES pls.

Surely, they must have also given you a phone number, or block of numbers.

Please post the INVITE for an incoming call, so we can see what format they use for the number, or if they use an account number or something else.

Sorry, I didn’t see your last post.

Try setting From User to
if that doesn’t work, try

If they all fail, you’ll need to look at an incoming call.

Thanks @shahin . Thanks everybody. Issue is solved.

You are welcome, i was told you something missing at your Outbound route side.

Well Done.

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