PBX Seems to be Declining (603) All SIP Calls

For a simple home communication system, I’ve set up some very simple SIP / Extensions. Go easy on me, I’m very new to this system.

For now the only way I’ve gotten them to work (in testing) is to take down the firewall. Still, I seem to be getting instant 603’s with every try from every phone. Why am I getting 603’s seemingly by default?

When I make a call, this is what it reports:

[code] <— SIP read from UDP:192.168.1.8:5060 —>
INVITE sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
Max-Forwards: 70
From: sip:0000FFFF004@192.168.1.6;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: sip:103@192.168.1.6
Contact: sip:0000FFFF004@192.168.1.8:5060
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Content-Type: application/sdp
Content-Length: 361

v=0
o=dinosaur 3611940779 0 IN IP4 192.168.1.8
s=sflphone
c=IN IP4 192.168.1.8
t=0 0
m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (12 headers 16 lines) ---
Sending to 192.168.1.8:5060 (NAT)
Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060

<--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as69cdb064
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5572b5df"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.8:5060 --->
ACK sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj43e20551-6dbc-411b-919a-ab117c06ae05
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as69cdb064
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6702 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.8:5060 --->
INVITE sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>
Contact: <sip:0000FFFF004@192.168.1.8:5060>
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Allow: PRACK, SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER, OPTIONS, MESSAGE
upported: replaces, 100rel
Authorization: Digest username="0000FFFF004", realm="asterisk", nonce="5572b5df", uri="sip:103@192.168.1.6", response="44810c7fbf0d8a99e34ea07b5e62ee79", algorithm=MD5
Content-Type: application/sdp
Content-Length: 361

v=0
o=dinosaur 3611940779 0 IN IP4 192.168.1.8
s=sflphone
c=IN IP4 192.168.1.8
t=0 0
m=audio 37600 RTP/AVP 0 3 8 9 110 111 112
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:111 speex/16000
a=rtpmap:112 speex/32000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 16 lines) ---
Sending to 192.168.1.8:5060 (NAT)
Using INVITE request as basis request - 16962f1e-d2e0-4987-af02-dc765ffa793f
Found peer '0000FFFF004' for '0000FFFF004' from 192.168.1.8:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 110
Found RTP audio format 111
Found RTP audio format 112
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format speex for ID 110
Found audio description format speex for ID 111
Found unknown media description format speex for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0x100e (gsm|ulaw|alaw|g722), peer - audio=0x20000120e (gsm|ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100e (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.8:37600
Looking for 103 in LocalSets (domain 192.168.1.6)
list_route: hop: <sip:0000FFFF004@192.168.1.8:5060>

<--- Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:103@192.168.1.6:5060>
Content-Length: 0


<------------>
    -- Executing [103@LocalSets:1] Dial("SIP/0000FFFF004-0000001a", "0000FFFF005") in new stack
  == Spawn extension (LocalSets, 103, 1) exited non-zero on 'SIP/0000FFFF004-0000001a'
Scheduling destruction of SIP dialog '16962f1e-d2e0-4987-af02-dc765ffa793f' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.8:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d;received=192.168.1.8;rport=5060
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as165ecdc9
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 INVITE
Server: Asterisk PBX SVN-branch-1.8-r416150
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.8:5060 --->
ACK sip:103@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;rport;branch=z9hG4bKPj7504ac55-fcbe-470a-a5c5-2174a0699d0d
Max-Forwards: 70
From: <sip:0000FFFF004@192.168.1.6>;tag=12e33b02-c7c1-4b83-9ba4-f08c426ece25
To: <sip:103@192.168.1.6>;tag=as165ecdc9
Call-ID: 16962f1e-d2e0-4987-af02-dc765ffa793f
CSeq: 6703 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.5:63992 --->

<------------->
Really destroying SIP dialog 'cb7123d1-4244-4673-a200-dc851e1c8415' Method: REGISTER[/code]

The phones themselves are not set to decline calls, so I can only assume its happening somewhere in Asterisk.

The parameter of Dial is invalid; it is missing a technology/ prefix. That is either totally invalid, or will be interpreted as Local/. I’d have to check the code to be sure.