Incoming SIP Calls SIP/2.0 404 Not Found

Asterisk Asterisk Support
1

So, I have a wholesale provider. I use them on of my VOIP servers - Trixbox v2.8.0.4, but its getting pretty old, So I put up a new box for VOIP, I can make outgoing calls no problem, I am having a problem receiving calls, The provider says my server is giving them a 404 error. I have allowed anonymous SIP calls turn on. But for some reason its still rejecting calls.

It appears that its not even hitting asterisk at all. When I call the provider, they do a call trace and they say, “We are getting a 404 from your Trixbox.”, “That usually means anonymous sip calls is not turned on.”

so I told them, "It is turned on, and it appears in the CLI that its not even coming through."
Then they tell me, "Well its going to your IP and your IP is giving us a 404 error"
I have it setup the exact same way as my other box is setup, but this one just does not work like its suppose to. Only difference between the two is the IP.

Any help will be much appreciated

2

  1. Turning on anonymous SIP is normally a very bad idea, unless you have a very good firewall configuration.

  2. Without a SIP trace of the failed exchange and the contents of your sip.conf (or users.conf), and exensions.conf, it is rather difficult to guess what is wrong.

3

Below is the trace from my SIP provider. I have changed the IP and telephone numbers.

INVITE sip:441618206000@m.e164.org.uk:5060 SIP/2.0
Via: SIP/2.0/UDP  213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>
Date: Wed, 23 Feb 2011 06:05:40 gmt
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
Supported: timer,replaces
Min-SE:  1800
Cisco-Guid: 3330977092-1044976096-2343763975-240685966
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 15
Remote-Party-ID: <sip:07855111257@213.166.5.133>;party=calling;screen=yes;privacy=off
Timestamp: 1298441140
Contact: <sip:07855111257@213.166.5.133:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 404

v=0
o=CiscoSystemsSIP-GW-UserAgent 2617 3375 IN IP4 213.166.5.133
s=SIP Call
c=IN IP4 213.166.5.133
t=0 0
m=audio 19016 RTP/AVP 8 18 4 3 2 0 101
c=IN IP4 213.166.5.133
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP  213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
CSeq: 101 INVITE
Server: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

INVITE sip:01618206111@217.41.3.11 SIP/2.0
Record-Route: <sip:213.166.5.129;lr=on;ftag=CC528F34-143F>
Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKd82f.04407d3.0
Via: SIP/2.0/UDP  213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: "07855111257" <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>
Date: Wed, 23 Feb 2011 06:05:40 gmt
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
Supported: timer,replaces
Min-SE:  1800
User-Agent: MSSGW
Allow: INVITE, BYE, CANCEL, ACK
CSeq: 101 INVITE
Max-Forwards: 14
Remote-Party-ID: <sip:07855111257@213.166.5.133>;party=calling;screen=yes;privacy=off
Timestamp: 1298441140
Contact: <sip:07855111257@213.166.5.133:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 404

v=0
o=CiscoSystemsSIP-GW-UserAgent 2617 3375 IN IP4 213.166.5.133
s=SIP Call
c=IN IP4 213.166.5.133
t=0 0
m=audio 19016 RTP/AVP 8 18 4 3 2 0 101
c=IN IP4 213.166.5.133
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKd82f.04407d3.0
Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: "07855111257" <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
CSeq: 101 INVITE
Content-Length: 0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKd82f.04407d3.0
Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: "07855111257" <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>;tag=efM-7276
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
CSeq: 101 INVITE
Content-Length: 0

ACK sip:01618206111@217.41.3.11 SIP/2.0
Via: SIP/2.0/UDP 213.166.5.129;branch=z9hG4bKd82f.04407d3.0
From: <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
To: <sip:441618206000@m.e164.org.uk>;tag=efM-7276
CSeq: 101 ACK
User-Agent: OpenSER (1.2.1-notls (i386/linux))
Content-Length: 0

SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: "07855111257" <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>;tag=efM-7276
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
CSeq: 101 INVITE
Content-Length: 0

ACK sip:441618206000@m.e164.org.uk:5060 SIP/2.0
Via: SIP/2.0/UDP  213.166.5.133:5060;branch=z9hG4bKC6AB43829
From: <sip:07855111257@213.166.5.133>;tag=CC528F34-143F
To: <sip:441618206000@m.e164.org.uk>;tag=efM-7276
Date: Wed, 23 Feb 2011 06:05:40 gmt
Call-ID: C68B4A05-3E4911E0-8F04D6A7-8B5D6257@213.166.5.133
Max-Forwards: 15
CSeq: 101 ACK
Content-Length: 0