[asterisk-users] SIP 603 Declined error message

Paul Belanger paul.belanger at polybeacon.com
Wed Feb 26 10:14:01 CST 2014 

On Wed, Feb 26, 2014 at 8:10 AM, Haley,Scott A
<scott.haley at edwardjones.com> wrote:
> 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) ---
>

You'll want to talk to the FreePBX guys, as you are just hanging up
the outbound call.

    -- 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

-- 
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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