--- Failed to authenticate on INVITE

Asterisk Asterisk SIP
1

Hello,

I meet some trouble when I try to register a SIP trunk to a domain (keyyo.net) to my Asterisk server.
Dialplan : “+33999999999” rings SIP account “test”
When the SIP trunk is registred to a specific IP, address everything works well (sound in both way when I call +33999999999).

But when I “sip reload” and the registrar change :
[Aug 23 10:48:30] – <SIP/27.b2bua.sip.internal-0000038e>AGI Script agi://127.0.0.1:4573/dial completed, returning 0
[Aug 23 10:48:30] – Executing [s@macro-dfimutu-dial:23] ExecIf(“SIP/27.b2bua.sip.internal-0000038e”, “1?Dial(SIP/test,20,tkKF(all-dial-hangup,s,1),)”) in new stack
[Aug 23 10:48:30] == Using SIP RTP TOS bits 184
[Aug 23 10:48:30] == Using SIP RTP CoS mark 5
[Aug 23 10:48:30] – Called SIP/test
[2021-08-23 10:48:30] NOTICE[4236][C-00003540]: chan_sip.c:25469 handle_response_invite: Failed to authenticate on INVITE to 'sip:33999999999@keyyo.net;tag=as243d4c04’
[Aug 23 10:48:33] – SIP/test-0000038f is ringing

So it works randomly.
What can I do?
On my sip.conf :
srvlookup = yes

Thx,

2

Registrars authenticate REGISTER, not INVITE. What do you mean by a change in registrar?

Also please provide the sip set debug on output for the failing transaction, and a working one.

3

As the registrar of the SIP trunk is a domain, my SIP trunk is registred to random IP addresses.
Incoming call works correctly only with one.

4

Your failing call is an outgoing one. Regarding multiple incoming addresses, the legacy chan_sip doesn’t cater for those well, but chan_pjsip has better support.

Whilst it is possible that the service provider requires a registration before it will accept INVITEs, that is a local policy decision. SIP does not require the use of a registrar for INVITEs.

5

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