Error in receiving/outgoing calls!

[Mercy1885]

Registration logs says "Trunk L:10000(PHONE.COM) has changed status to registered"

Second logs" The following service(s) were interrupted: 3CXAudioProvider01 3CXSystemService01 3CXIVR01 3CXCallFlow01 3CXGatewayService 3CXQueueManager01 3CXPhoneSystemMC01 3CXPhoneSystem01 3CXCfgServ01"

Incoming call error logs" Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:[email protected]:5060;transport=udp;jtr=4788-3908 SIP/2.0 Via: SIP/2.0/UDP 72.1.46.10:5060;branch=z9hG4bK1sansay2277411539rdb71127 Record-Route: <sip:[email protected]:5060;lr;transport=udp> Contact: <sip:+1XXXXXX @72.1.46.10:5060> To: <sip:[email protected]> From: "BOXXXXJ" <sip:[email protected]>;tag=sansay2277411539rdb71127 Call-ID: cc6dfc11-8bbb-123e-6c98-d4bed9a8ed9b CSeq: 1 INVITE Session-Expires: 600;refresher=uac Min-SE: 90 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Content-Type: application/sdp Supported: timer, replaces, path X-Linked-UUID: dce50bfa-f135-4a38-89ac-4cffbddd28fc X-Voip-Id: 3340787 Content-Length: 457 v=0 o=Sansay-VSXi 188 1 IN IP4 72.1.46.10 s=Session Controller c=IN IP4 72.1.47.173 t=0 0 m=audio 30134 RTP/AVP 102 9 0 8 103 101 a=rtpmap:102 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:102 useinbandfec=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=fmtp:103 0-15 a=fmtp:101 0-15 a=ptime:20

Outgoing Call log "

Call or Registration to +187XXXXXX@([email protected]) has failed. sip:72.1.46.10:5060;lr replied: Not Found (404)"

 

[NikosT_3CX]

@Mercy1885 as you may see from the message you have posted, the 404 Not Found comes from IP 72.1.46.10.
Most probably this is your provider 's IP, so give them a call and ask them why they give this message to the outbound call of the PBX :
Call or Registration to +187XXXXXX@([email protected]) has failed. sip:72.1.46.10:5060;lr replied: Not Found (404)"

For the Incoming calls, ensure that you have configured the DID as the provider sends it on the TO ( To: <sip:998130 ) and you have it assigned to its destination, like extension or IVR or Queue or Ring Group.

 

[Mercy1885]

Under ring group I created ring group and add the user1, same DID# assigned to the user1 still for inbound seeing the same error log "Unidentified Incoming Call. Review INVITE and adjust source identification: INVITE sip:[email protected]:5060;transport=udp;jtr=3218-3744 SIP/2.0 Via: SIP/2.0/UDP 72.1.46.10:5060;branch=z9hG4bK1sansay2277812574rdb45141 Record-Route: <sip:[email protected]:5060;lr;transport=udp> Contact: <sip:[email protected]:5060> To: <sip:[email protected]> From: "BOUND BROOK NJ" <sip:[email protected]>;tag=sansay2277812574rdb45141 Call-ID: f4544f8f-8bc4-123e-a9b7-00219b9e3ef4 CSeq: 1 INVITE Session-Expires: 600;refresher=uac Min-SE: 90 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Content-Type: application/sdp Supported: timer, replaces, path X-Linked-UUID: d3f6d99b-79ec-421e-a6b4-5c7865b4cf41 X-Voip-Id: 3340787 Content-Length: 464 v=0 o=Sansay-VSXi 188 1 IN IP4 72.1.46.10 s=Session Controller c=IN IP4 72.1.47.174 t=0 0 m=audio 36734 RTP/AVP 102 9 0 8 103 101 a=rtpmap:102 opus/48000/2 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:103 telephone-event/48000 a=rtpmap:101 telephone-event/8000 a=fmtp:102 useinbandfec=1; cbr=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40 a=fmtp:103 0-15 a=fmtp:101 0-15 a=ptime:20"

 

[Mercy1885]

Addition to above, in voice and chat default route to ring group and selected the ring group

 

[NikosT_3CX]

Did you add the DID 998130 on the trunk and then assign it to the Ring Group ?

 

[Mercy1885]

I add TFN#

 

[Mercy1885]

Did you add the DID 998130 on the trunk and then assign it to the Ring Group ?

So main trunk will be 998130?

 

[NikosT_3CX]

Dont know if this is the Main Trunk or not, but by the time the Provider sends this number on the TO section, then
this must exist at least as a DID on the PBX side and assigned somewhere.

 

[Mercy1885]

Finally Inbound fixed!
And outbound failed "Call or Registration to +187784XXXX0@([email protected]) has failed. sip:72.1.46.10:5060;lr replied: Not Found (404)"

 

[NikosT_3CX]

as you may see from the message you have posted, the 404 Not Found comes from IP 72.1.46.10.
Most probably this is your provider 's IP, so give them a call and ask them why they give this message to the outbound call of the PBX :

"Call or Registration to +187784XXXX0@([email protected]) has failed. sip:72.1.46.10:5060;lr replied: Not Found (404)"