In case it helps, here's the invite my Asterisk system sends to the ITSP (obfuscated a bit). This should be triggering a 407 from the ITSP but it's not. So I must be missing something in this message...can't see what
<--- Transmitting SIP request (930 bytes) to UDP:54.172.60.0:5060 ---> INVITE sip:12223334444@54.172.60.0:5060 SIP/2.0 Via: SIP/2.0/UDP 122.59.105.83:5060;rport;branch=z9hG4bKPj1b1875dc-11b7-4882-bbe3-d56c6041043a From: "MYNAME" <sip:16667778888@192.168.253.4>;tag=d147259b-dc0a-454e-8c6c-14ac59e85197 To: <sip:12223334444@54.172.60.0> Contact: <sip:asterisk@122.59.105.83:5060> Call-ID: db46e226-73de-46f9-8b96-388eb5f0dd5e CSeq: 13035 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: MyUA Content-Type: application/sdp Content-Length: 235 v=0 o=- 954636103 954636103 IN IP4 122.59.105.83 s=Asterisk c=IN IP4 122.59.105.83 t=0 0 m=audio 15860 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv -----Original Message----- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of TTT Sent: Wednesday, June 21, 2023 1:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com> Subject: Re: [asterisk-users] PJSIP not performing outbound authentication I didn't use pjsip_wizard, I'm kind of crafting this by hand as I learn. I actually have a plain asterisk, and a FreePBX, system to help me learn. I sometimes create something in FreePBX to see what it does to the config files. So that's how I modelled my pjsip.X.conf files If I issue the command "pjsip show endpoint Twilio" it does show that outbound_auth=Twilio Does that mean the initial invite will contain authentication info? Or does Asterisk still wait for a 407?? (I'm wondering if maybe Asterisk is working normally, this is a Twilio config problem). And I confirmed the CID info matches an account on Twilio, so it's not that. -----Original Message----- From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Henning Follmann Sent: Wednesday, June 21, 2023 1:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PJSIP not performing outbound authentication On Wed, Jun 21, 2023 at 05:19:11PM +0000, TTT wrote: > I am using Asterisk 20.3.0 with PJSIP. I have setup a trunk to my ISP > (Twilio) who requires outbound authentication. My pjsip.auth.conf contains: > > [Twilio] > type=auth > auth_type=userpass > password=mysecret > username=myun > > However, my calls using the trunk are rejected with a 403. Using pjsip > logging I notice that the outgoing invite does not have an > authentication line. Why is Asterisk not sending credentials to the > ISP? SIP transactions > are: > > INVITE > < 100 TRYING > < 403 FORBIDDEN > > Or is this normal? Must Twilio respond with a 407 which will cause > Asterisk to authenticate? > > Twilio has a nice technical document to setup a trunk with PJSIP. It includes an example for a pjsip_wizard.conf https://assets.cdn.prod.twilio.com/documents/TwilioElasticSIPTrunking-AsteriskPBX-Configuration-Guide-Version2-1-FINAL-09012018.pdf Maybe that helps. And make sure for your outgoing calls to set the callerid to a valid caller Id which ist authorized with your twilio account. It will not allow outgoing calls if the number is not recognized by twilio -H -- Henning Follmann | hfollm...@itcfollmann.com -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
