404 not found when making south-bound call

General Help
1

I can make north-bound no problem.
However I cannot make a south-bound call with 404 not found.
Anyone has any idea what might be the cause.

pbxuser2011

2

How are east and west bound calls working?

Perhaps you need to reset your directional gyro?

3

I have provided the PEER info and traces for this 404 not-found error.

Anyone have any idea will be great.

Thanks,
pbxuser2011

host=sippbx.iot
type=peer
dtmfmode=rfc2833
allow=all
username=9192211200
secret=123456
fromuser=9192211200
fromdomain=sippbx.iot

<— SIP read from UDP:172.28.245.4:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Max-Forwards: 19
Session-Expires: 3600;refresher=uas
Min-SE: 360
Supported: timer, 100rel
To: sip:[email protected]:10000
From: “SIPUSER 6002” sip:[email protected];tag=3530899839-82336
Remote-Party-Id: sip:[email protected];party=called
Call-ID: [email protected]
CSeq: 1 INVITE
Allow: UPDATE
Via: SIP/2.0/UDP 172.28.245.4:5060;branch=z9hG4bKde63f4f6d9e90d7fa3b99e26f3d6dee3
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Accept: application/sdp
User-Agent: Nortel SESM 12.0.6.16
Content-Length: 288

v=0
o=IOTMSX1-1 8000 8000 IN IP4 172.28.245.4
s=sip call
c=IN IP4 172.28.245.5
t=0 0
m=audio 11806 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (17 headers 14 lines) —
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 172.28.245.4 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘sippbx.iot’ for ‘9199926002’ from 172.28.245.4:5060
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.28.245.5:11806
Looking for 9192211200 in from-trunk-sip-sippbx.iot (domain 47.135.41.119)

<— Reliably Transmitting (no NAT) to 172.28.245.4:5060 —>
SIP/2.0 404 Not Found <-------- 404
Via: SIP/2.0/UDP 172.28.245.4:5060;branch=z9hG4bKde63f4f6d9e90d7fa3b99e26f3d6dee3;received=172.28.245.4
From: “SIPUSER 6002” sip:[email protected];tag=3530899839-82336
To: sip:[email protected]:10000;tag=as3b083012
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.11
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘[email protected]’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:172.28.245.4:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Max-Forwards: 70
To: sip:[email protected]:10000;tag=as3b083012
From: “SIPUSER 6002” sip:[email protected];tag=3530899839-82336
Call-ID: [email protected]
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.28.245.4:5060;branch=z9hG4bKde63f4f6d9e90d7fa3b99e26f3d6dee3
Contact: sip:[email protected]:5060
Content-Length: 0

<------------->

4

Contact your carrier and see why they are rejecting the calls.