Reliably Transmitting (NAT) SIP/2.0 404 Not Found

Asterisk Asterisk Support
1

Hello,

I have troubles with some incoming calls. Most of the time i’m just available from the outside, but sometimes i get a busy tone (I check my asterisk regularly through the day). When incoming calls go bad, the following in my asterisk -r sip set debug output is different from a succesfull incoming call:

Succesfull incoming call:

<--- Transmitting (NAT) to 81.23.228.129:5060 --->
SIP/2.0 100 Trying

Busy tone/failed incoming call

<--- Reliably Transmitting (NAT) to 81.23.228.150:5060 --->
SIP/2.0 404 Not Found

Complete output of incoming call going bad:

Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 1.4.21.2 currently running on elastix (pid = 2288)
Verbosity is at least 3
elastix*CLI> exit
<--- SIP read from 81.23.228.150:5060 --->
INVITE sip:314571123**@84.25.171.*** SIP/2.0
Record-Route: <sip:81.23.228.150;lr=on;ftag=606110252;fcd=yes;did=1d3.d266f5f5>
Record-Route: <sip:81.23.228.129;lr=on;ftag=606110252;fcd=yes;did=1d3.178f73e>
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK1ff9.32a5ef7.0
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK1ff9.35a62dd4.0
Via: SIP/2.0/UDP 80.252.84.190:5060;branch=z9hG4bK1833169811
From: "0455719101" <sip:0455719101@80.252.84.190>;tag=606110252
To: <sip:314571123**@budgetphone.nl:5060>
Call-ID: 1526761710@80.252.84.190
CSeq: 20 INVITE
Contact: <sip:SIP_5Fd@80.252.84.190>
Max-Forwards:  8
Allow: INVITE, ACK, BYE, CANCEL, INFO, UPDATE, REGISTER
Content-Type: application/sdp
Content-Length: 297

v=0
o=SIP_5Fd 123456 654321 IN IP4 80.252.84.190
s=-
c=IN IP4 85.17.186.6
t=0 0
m=audio 62988 RTP/AVP 3 4 8 0 18 101
a=rtpmap:3 GSM/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-15

<------------->
--- (15 headers 13 lines) ---
Sending to 81.23.228.150 : 5060 (NAT)
Using INVITE request as basis request - 1526761710@80.252.84.190
Found peer 'BUDGETPHONE'
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 85.17.186.6:62988
Found audio description format GSM for ID 3
Found audio description format G723 for ID 4
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 85.17.186.6:62988
Looking for 314571123** in from-budgetphone (domain 84.25.171.***)

<--- Reliably Transmitting (NAT) to 81.23.228.150:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK1ff9.32a5ef7.0;received=81.23.228.150
Via: SIP/2.0/UDP 81.23.228.129;branch=z9hG4bK1ff9.35a62dd4.0
Via: SIP/2.0/UDP 80.252.84.190:5060;branch=z9hG4bK1833169811
From: "0455719101" <sip:0455719101@80.252.84.190>;tag=606110252
To: <sip:314571123**@budgetphone.nl:5060>;tag=as5a5e8709
Call-ID: 1526761710@80.252.84.190
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1526761710@80.252.84.190' in 6400 ms (Method: INVITE)
elastix*CLI> exit
<--- SIP read from 81.23.228.150:5060 --->
ACK sip:314571123**@84.25.171.*** SIP/2.0
Via: SIP/2.0/UDP 81.23.228.150;branch=z9hG4bK1ff9.32a5ef7.0
From: "0455719101" <sip:0455719101@80.252.84.190>;tag=606110252
Call-ID: 1526761710@80.252.84.190
To: <sip:314571123**@budgetphone.nl:5060>;tag=as5a5e8709
CSeq: 20 ACK
Max-Forwards: 70
User-Agent: OpenSER (1.3.2-tls (i386/linux))
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '1526761710@80.252.84.190' Method: ACK

Thank you very much for trying to help me with solving my problem (making me nuts for about a week now :laughing: :unamused: )