[asterisk-users] PJSIP not performing outbound authentication

TTT lists at telium.io
Wed Jun 21 13:11:04 CDT 2023


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 at 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 at 192.168.253.4>;tag=d147259b-dc0a-454e-8c6c-14ac59e85197
To: <sip:12223334444 at 54.172.60.0>
Contact: <sip:asterisk at 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-bounces at 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 at 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-bounces at lists.digium.com] On Behalf Of Henning Follmann
Sent: Wednesday, June 21, 2023 1:31 PM
To: asterisk-users at 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           | hfollmann at itcfollmann.com


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Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
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