SIP/2.0 480 No Routes Found

[jop.is.my.name]

Hi Everyone,

I'm currently experiencing a problem with my 3CX server. I had setup a generic trunk in 3CX connecting to our SIP Provider since it is not supported in 3CX options. The scenario is that I can call the pilot number (DID) of the SIP but when I'm trying to make an outbound call using the SIP trunk, my call cannot pushed through. I captured the traffic using wireshark and found that it displays the SIP/2.0 480 No Routes Found in the capture.

Setup:
3CX server is installed in a Windows Server operating system. I added persistent route to our SIP provider's SBC IP. SBC can be pinged in the router and in the Windows Operating system.

What can I do to resolved this? Thanks in advance.

-------------------------------------------------------------------------------
TRAFFIC CAPTURE
-------------------------------------------------------------------------------
INVITE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 172.16.XX.X:5060;branch=z9hG4bK-524287-1---8c382e40a551742d;rport

Max-Forwards: 70

Contact: <sip:[email protected]:5060>

To: <sip:[email protected]>

From: "091XXXXXX"<sip:[email protected]>;tag=0e0de832

Call-ID: GjCojvGFgGA-AlK-FEbIEw..

CSeq: 1 INVITE

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

Content-Type: application/sdp

Supported: replaces, timer

User-Agent: 3CXPhoneSystem 16.0.5.612 (612)

Remote-Party-ID: "091XXXXXXX"<sip:[email protected]>;party=calling

Content-Length: 276



v=0

o=3cxPS 340141277184 158108483585 IN IP4 172.16.XX.X

s=3cxPS Audio call

c=IN IP4 172.16.XX.X

t=0 0

m=audio 8116 RTP/AVP 0 8 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=sendrecv

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 172.16.XX.X:5060;received=172.16.XX.X;branch=z9hG4bK-524287-1---8c382e40a551742d;rport=5060

To: <sip:[email protected]>

From: "09190580500"<sip:[email protected]>;tag=0e0de832

Call-ID: GjCojvGFgGA-AlK-FEbIEw..

CSeq: 1 INVITE



SIP/2.0 480 No Routes Found

Via: SIP/2.0/UDP 172.16.XX.X:5060;received=172.16.XX.X;branch=z9hG4bK-524287-1---8c382e40a551742d;rport=5060

To: <sip:[email protected]>;tag=aprqngfrt-llom7c2000020

From: "091XXXXXXXX"<sip:[email protected]>;tag=0e0de832

Call-ID: GjCojvGFgGA-AlK-FEbIEw..

CSeq: 1 INVITE

 

[leejor]

I'm going to assume that you have not created an outbound rule that will handle dialled number 091XXXXXXXX.

https://www.3cx.com/docs/manual/sip-trunks/

 

[jop.is.my.name]

I have outbound rule for this. I have other SIP route as well, Other SIP Trunk can connect the call successfully using the same outbound rule. The new one that I have is a bit weird. I can tracert to the SBC and even ping the IP, but the call cannot pushed through.

 

[NickD_3CX]

What does your SIP Trunk provider say? I think you should ask them to check if the SIP Packet is reaching their SBC.
They could then give you a clue why the call is failing.