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Solved 603 Decline with 3CX v18 RC and previous betas for internal calls

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webstean

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I've found that internal extensions don't work with v18 RC and a couple of beta before that. External calls work fine.
I'm getting 603 Decline on any internal call - I'm using SBC accross three sites, with a cloud hosted 3CX.
It the same for a native SIP client and 3CX iOS soft phone and Fanvil handsets.

3CX Activity Log:-

Code: 
08/03/2021 10:46:46 AM - [CM502001]: Source info: From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392; To: <sip:666@FQDN;transport=udp>
08/03/2021 10:46:46 AM - [CM503013]: Call(C:7): Incoming call rejected, caller is unknown; msg=Invite-IN Recv Req INVITE from 1.136.111.201:48081 tid=619b0f961a0d00ec Call-ID=e2beb1e22b696e1b: INVITE sip:666@FQDN;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK619b0f961a0d00ec;rport=48081;received=1.136.111.201 Max-Forwards: 70 Contact: <sip:[email protected]:33385> To: <sip:666@FQDN;transport=udp> From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392 Call-ID: e2beb1e22b696e1b CSeq: 16275 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, SUBSCRIBE, INFO, MESSAGE, REFER Content-Type: application/sdp Proxy-Authorization: Digest username="1Q0FM8Asa9",realm="3CXPhoneSystem",nonce="414d5359610891f523:6d8c98f001af782164dea04f9d1f3196",uri="sip:666@FQDN;transport=udp",response="6c4da98fbbcfea282f1d80978ce5a006",algorithm=MD5 Supported: gruu User-Agent: baresip v1.1.0 (x86_64/linux) Content-Length: 352 v=0 o=- 690483025 1542754858 IN IP4 172.24.175.206 s=- c=IN IP4 172.24.175.206 t=0 0 m=audio 12584 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=label:1 a=rtcp-rsize a=ssrc:4190757915 cname:sip:667@FQDN a=minptime:20 a=ptime:20
08/03/2021 10:46:46 AM - Leg L:7.1[Unknown:] is terminated: Cause: BYE from PBX
08/03/2021 10:46:45 AM - [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification: Invite-UNK Recv Req INVITE from 1.136.111.201:48081 tid=29e508bb2136392f Call-ID=e2beb1e22b696e1b: INVITE sip:666@FQDN;transport=udp SIP/2.0 Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK29e508bb2136392f;rport=48081;received=1.136.111.201 Max-Forwards: 70 Contact: <sip:[email protected]:33385> To: <sip:666@FQDN;transport=udp> From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392 Call-ID: e2beb1e22b696e1b CSeq: 16274 INVITE Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, NOTIFY, SUBSCRIBE, INFO, MESSAGE, REFER Content-Type: application/sdp Supported: gruu User-Agent: baresip v1.1.0 (x86_64/linux) Content-Length: 352 v=0 o=- 690483025 1542754858 IN IP4 172.24.175.206 s=- c=IN IP4 172.24.175.206 t=0 0 m=audio 12584 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=label:1 a=rtcp-rsize a=ssrc:4190757915 cname:sip:667@FQDN a=minptime:20 a=ptime:20
08/03/2021 10:46:43 AM - Currently active calls [none]

SIP Trace:-


Code: 
667@FQDN: selected for request

call uri: sip:666@FQDN

ua: using best effort AF: af=AF_INET

call: connecting to 'sip:666@FQDN;transport=udp'..

call id: e2beb1e22b696e1b

#

UDP 172.24.175.206:33385 -> IP:PORT

INVITE sip:666@FQDN;transport=udp SIP/2.0

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK29e508bb2136392f;rport

Contact: <sip:[email protected]:33385>

Max-Forwards: 70

To: <sip:666@FQDN;transport=udp>

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16274 INVITE

User-Agent: baresip v1.1.0 (x86_64/linux)

Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER

Supported: gruu

Content-Type: application/sdp

Content-Length: 352



v=0

o=- 690483025 1542754858 IN IP4 172.24.175.206

s=-

c=IN IP4 172.24.175.206

t=0 0

m=audio 12584 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=label:1

a=rtcp-rsize

a=ssrc:4190757915 cname:sip:667@FQDN

a=minptime:20

a=ptime:20



#

UDP IP:PORT -> 172.24.175.206:33385

SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK29e508bb2136392f;rport=48081;received=1.136.111.201

Proxy-Authenticate: Digest nonce="414d5359610891f523:6d8c98f001af782164dea04f9d1f3196",algorithm=MD5,realm="3CXPhoneSystem"

To: <sip:666@FQDN;transport=udp>;tag=42dfcd5e

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16274 INVITE

Content-Length: 0





#

UDP 172.24.175.206:33385 -> IP:PORT

ACK sip:666@FQDN;transport=udp SIP/2.0

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK29e508bb2136392f;rport

Max-Forwards: 70

To: <sip:666@FQDN;transport=udp>;tag=42dfcd5e

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16274 ACK

User-Agent: baresip v1.1.0 (x86_64/linux)

Content-Length: 0





#

UDP 172.24.175.206:33385 -> IP:PORT

INVITE sip:666@FQDN;transport=udp SIP/2.0

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK619b0f961a0d00ec;rport

Contact: <sip:[email protected]:33385>

Max-Forwards: 70

Proxy-Authorization: Digest username="1Q0FM8Asa9", realm="3CXPhoneSystem", nonce="414d5359610891f523:6d8c98f001af782164dea04f9d1f3196", uri="sip:666@FQDN;transport=udp", response="6c4da98fbbcfea282f1d80978ce5a006", algorithm=MD5

To: <sip:666@FQDN;transport=udp>

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16275 INVITE

User-Agent: baresip v1.1.0 (x86_64/linux)

Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY,SUBSCRIBE,INFO,MESSAGE,REFER

Supported: gruu

Content-Type: application/sdp

Content-Length: 352



v=0

o=- 690483025 1542754858 IN IP4 172.24.175.206

s=-

c=IN IP4 172.24.175.206

t=0 0

m=audio 12584 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=sendrecv

a=label:1

a=rtcp-rsize

a=ssrc:4190757915 cname:sip:667@FQDN

a=minptime:20

a=ptime:20



#

UDP IP:PORT -> 172.24.175.206:33385

SIP/2.0 603 Decline

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK619b0f961a0d00ec;rport=48081;received=1.136.111.201

To: <sip:666@FQDN;transport=udp>;tag=8f0b2c2a

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16275 INVITE

Warning: 499 vmcca87306dd.kw4km1relcxevmrqjkvutzlkgg.qx.internal.cloudapp.net "Caller is forbidden"

Content-Length: 0





#

UDP 172.24.175.206:33385 -> IP:PORT

ACK sip:666@FQDN;transport=udp SIP/2.0

Via: SIP/2.0/UDP 172.24.175.206:33385;branch=z9hG4bK619b0f961a0d00ec;rport

Max-Forwards: 70

To: <sip:666@FQDN;transport=udp>;tag=8f0b2c2a

From: <sip:667@FQDN>;tag=62d6e2fe7a4f1392

Call-ID: e2beb1e22b696e1b

CSeq: 16275 ACK

User-Agent: baresip v1.1.0 (x86_64/linux)

Content-Length: 0





sip:666@FQDN;transport=udp: session closed: 603 Decline
 
Please try the following and let me know what happens:

1. Access the Management Console and navigate to "Settings >> Network".
2. Click on the FQDN tab.
3. Make sure that the "Local SIP Domain" field(2) contains the PBXs FQDN exactly as show in the "Fully Qualified Domain Name" field above(1)
1627974909173.png

4. Click on OK and check to see if the issue persists.
 
  • Like
Reactions: webstean
Ok - Thanks - that fixes it.
 
  • Like
Reactions: ChrisC_3CX
Great! Glad this got sorted!
 
  • Like
Reactions: webstean
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