Solved SIP problem

[DaleS]

Hello again!

I have a python script that runs when the front gate is opened node-red runs this script and plays a wav file to a paging group.
This works well with asterisk however with 3cx I am getting this error

04/18/2022 5:09:07 PM - [CM102001]: Authentication failed for AuthFail Recv Req INVITE from 192.168.2.253:5080 tid=PjRppJ9634cNnWtkpwQ6bQiJc1wtofiSrK Call-ID=XQufRG39DSztec6dwFav1x0Kt1JOukoX:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.253:5080;rport=5080;branch=z9hG4bKPjRppJ9634cNnWtkpwQ6bQiJc1wtofiSrK
Max-Forwards: 70 Route: <sip:192.168.2.51;lr>
Contact: <sip:[email protected]:5080;ob>
To: sip:[email protected]
From: sip:420@gateopener;tag=6D-JSPKUzQt0WxSvkkCljjXyU8Q.oEc9
Call-ID: XQufRG39DSztec6dwFav1x0Kt1JOukoX
CSeq: 11543 INVITE
Session-Expires: 1800
Min-SE: 90
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Type: application/sdp
Proxy-Authorization: Digest username="421",realm="3CXPhoneSystem",nonce="414d5359625d2ab336:c8d0c8a93f0cad12a0dfef18f20d4e30",uri="sip:[email protected]",response="1ff120d470cd555a114b1af3ed9121e4",algorithm=MD5
Supported: replaces, 100rel, timer, norefersub
User-Agent: pjsip python Content-Length: 634

v=0
o=- 3859261747 3859261747 IN IP4 192.168.2.253
s=pjmedia
b=AS:84
t=0 0
a=X-nat:0
m=audio 41327 RTP/AVP 96 97 98 99 3 0 8 9 120 121 122
c=IN IP4 192.168.2.253
b=TIAS:64000
a=rtcp:41328 IN IP4 192.168.2.253
a=sendrecv
a=rtpmap:96 speex/16000
a=rtpmap:97 speex/8000
a=rtpmap:98 speex/32000
a=rtpmap:99 iLBC/8000
a=fmtp:99 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:120 telephone-event/16000
a=fmtp:120 0-16
a=rtpmap:121 telephone-event/8000
a=fmtp:121 0-16
a=rtpmap:122 telephone-event/32000
a=fmtp:122 0-16
a=ssrc:1783053897
cname:27893fe209ea780c
; Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings

The code is returning this

--end msg--
21:47:58.588 pjsua_core.c .RX 362 bytes Response msg 603/INVITE/cseq=12480 (rdata0xb4101504) from UDP 192.168.2.51:5060:
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 192.168.2.253:5080;rport=5080;branch=z9hG4bKPj.CV.CuMcCVhV3izu0yD5vZ.bbf55nz
To: <sip:[email protected]>;tag=133d064b
From: <sip:421@gateopener>;tag=OG9y8eM2WZ5JwfHUiB2EAnmrSOJFQCTB
Call-ID: mJYuwcoN9NBcNlcI.f23P3dTEGs-nkSL
CSeq: 12480 INVITE
Warning: 499 pbx3.numpty.local "Caller is forbidden"
Content-Length: 0

I have checked and I believe the password and username are correct.

Is there a way to allow calling without SIP registration?

Any Help appreciated

 

[YiannisH_3CX]

Hi @DaleS

Is the extension 420 registered normally in 3CX? I can see that the extension is 420 but the auth id is 421? Also the From header is missing the <> signs so not sure if this will work.

 

[DaleS]

Thanks for the reply.

I have ditched pjsua infavour of baresip and its working perfectly

Appreciate prompt reply for support!!

You can close the thread

 

[YiannisH_3CX]

Glad to see you were able to resolve your issue.