[asterisk-users] SIP 603 Declined error message

Wed Feb 26 07:10:34 CST 2014

I have a SIP trunk from my Asterisk server to an Avaya CM server. If I place calls inbound, everything works fine. If I place calls outbound, originating from the Asterisk box, everything works fine (I have done this with the use of the .call files). If I setup an extension with the findme-followme feature and have it try to hair-pin a call back out the same trunk to the Avaya, I get a "SIP/2.0 603 Declined" message. Here is the output.

Any reason that this might be happening? It has been working up until now this week. I rebooted the machine on Tuesday.

<--- SIP read from TCP:172.17.184.46:31285 --->
INVITE sip:51104 at edj.devjones.com SIP/2.0
From: "Haley, Scott" <sip:3145152244 at edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Max-Forwards: 71
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Supported: 100rel,histinfo,join,replaces,sdp-anat,timer
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,INFO,PRACK,PUBLISH,UPDATE
User-Agent: Avaya CM/R016x.02.0.823.0
Contact: "Haley, Scott" <sip:3145152244 at 172.17.184.46;transport=tcp>
Route: <sip:192.168.122.51;transport=tcp;lr;phase=terminating>
Accept-Language: en;q=1
Alert-Info: <cid:internal at edj.devjones.com>;avaya-cm-alert-type=internal
History-Info: <sip:51104 at edj.devjones.com>;index=1
History-Info: "51104" <sip:51104 at edj.devjones.com>;index=1.1
Min-SE: 1200
P-Asserted-Identity: "Haley, Scott" <sip:3145152244 at edwardjones.com>
Record-Route: <sip:172.17.184.46;transport=tcp;lr>
Session-Expires: 1200;refresher=uac
Content-Type: application/sdp
Content-Length: 257

v=0
o=- 1393419743 1 IN IP4 172.17.184.46
s=-
c=IN IP4 172.17.184.93
b=AS:64
t=0 0
a=avf:avc=n prio=n
a=csup:avf-v0
m=audio 28196 RTP/AVP 0 18 127
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000
<------------->
--- (23 headers 13 lines) ---
Sending to 172.17.184.46:31285 (NAT)
Using INVITE request as basis request - 8066eb6f589ce3125b652973b4b00
Found peer 'trunk503in' for '3145152244' from 172.17.184.46:31285
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 127
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 127
Capabilities: us - 0x100c (ulaw|alaw|g722), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing)
Peer audio RTP is at port 172.17.184.93:28196
Looking for 51104 in from-trunk-sip-trunk503out (domain edj.devjones.com)
list_route: hop: <sip:172.17.184.46;transport=tcp;lr>

<--- Transmitting (NAT) to 172.17.184.46:31285 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
Record-Route: <sip:172.17.184.46;transport=tcp;lr>
From: "Haley, Scott" <sip:3145152244 at edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1200;refresher=uac
Contact: <sip:51104 at 192.168.122.51:5060;transport=TCP>
Content-Length: 0

<------------>
    -- Executing [51104 at from-trunk-sip-trunk503out:1] Set("SIP/trunk503in-0000010b", "GROUP()=OUT_1") in new stack
    -- Executing [51104 at from-trunk-sip-trunk503out:2] Goto("SIP/trunk503in-0000010b", "from-trunk,51104,1") in new stack
    -- Goto (from-trunk,51104,1)
    -- Executing [51104 at from-trunk:1] Set("SIP/trunk503in-0000010b", "__FROM_DID=51104") in new stack
    -- Executing [51104 at from-trunk:2] Gosub("SIP/trunk503in-0000010b", "app-blacklist-check,s,1") in new stack
    -- Executing [s at app-blacklist-check:1] GotoIf("SIP/trunk503in-0000010b", "0?blacklisted") in new stack
    -- Executing [s at app-blacklist-check:2] Set("SIP/trunk503in-0000010b", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s at app-blacklist-check:3] Return("SIP/trunk503in-0000010b", "") in new stack
    -- Executing [51104 at from-trunk:3] Gosub("SIP/trunk503in-0000010b", "cidlookup,cidlookup_1,1") in new stack
    -- Executing [cidlookup_1 at cidlookup:1] GotoIf("SIP/trunk503in-0000010b", "1?cidlookup,cidlookup_return,1") in new stack
    -- Goto (cidlookup,cidlookup_return,1)
    -- Executing [cidlookup_return at cidlookup:1] ExecIf("SIP/trunk503in-0000010b", "0?Set(CALLERID(name)=)") in new stack
    -- Executing [cidlookup_return at cidlookup:2] Return("SIP/trunk503in-0000010b", "") in new stack
    -- Executing [51104 at from-trunk:4] ExecIf("SIP/trunk503in-0000010b", "0 ?Set(CALLERID(name)=3145152244)") in new stack
    -- Executing [51104 at from-trunk:5] Set("SIP/trunk503in-0000010b", "__CALLINGPRES_SV=allowed_not_screened") in new stack
   -- Executing [51104 at from-trunk:6] Set("SIP/trunk503in-0000010b", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [51104 at from-trunk:7] Goto("SIP/trunk503in-0000010b", "app-blackhole,hangup,1") in new stack
    -- Goto (app-blackhole,hangup,1)
    -- Executing [hangup at app-blackhole:1] NoOp("SIP/trunk503in-0000010b", "Blackhole Dest: Hangup") in new stack
    -- Executing [hangup at app-blackhole:2] Hangup("SIP/trunk503in-0000010b", "") in new stack
  == Spawn extension (app-blackhole, hangup, 2) exited non-zero on 'SIP/trunk503in-0000010b'
Scheduling destruction of SIP dialog '8066eb6f589ce3125b652973b4b00' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 172.17.184.46:31285 --->
SIP/2.0 603 Declined
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
Via: SIP/2.0/TCP 172.18.78.67;branch=z9hG4bK8066eb6f589ce3126b652973b4b00
From: "Haley, Scott" <sip:3145152244 at edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>;tag=as06e2e068
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 INVITE
Server: FPBX-2.8.1(1.8.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

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

<--- SIP read from TCP:172.17.184.46:31285 --->
ACK sip:51104 at edj.devjones.com SIP/2.0
From: "Haley, Scott" <sip:3145152244 at edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104 at edj.devjones.com>;tag=as06e2e068
Call-ID: 8066eb6f589ce3125b652973b4b00
CSeq: 1 ACK
Max-Forwards: 70
Via: SIP/2.0/TCP 172.17.184.46;branch=z9hG4bK8066eb6f589ce3126b652973b4b00;received=172.17.184.46;rport=31285
User-Agent: Avaya CM/R016x.02.0.823.0
Route: <sip:192.168.122.51;transport=tcp;lr;phase=terminating>
Content-Length: 0

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

Thanks,
Scott Haley
Edward Jones Investments

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