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scott.haley at edwardj... Guest
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Posted: Wed Feb 26, 2014 8:11 am Post subject: [asterisk-users] SIP 603 Declined error message |
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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@edj.devjones.com SIP/2.0
From: "Haley, Scott" <sip:3145152244@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@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@edj.devjones.com>;avaya-cm-alert-type=internal
History-Info: <sip:51104@edj.devjones.com>;index=1
History-Info: "51104" <sip:51104@edj.devjones.com>;index=1.1
Min-SE: 1200
P-Asserted-Identity: "Haley, Scott" <sip:3145152244@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@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@192.168.122.51:5060;transport=TCP>
Content-Length: 0
<------------>
-- Executing [51104@from-trunk-sip-trunk503out:1] Set("SIP/trunk503in-0000010b", "GROUP()=OUT_1") in new stack
-- Executing [51104@from-trunk-sip-trunk503out:2] Goto("SIP/trunk503in-0000010b", "from-trunk,51104,1") in new stack
-- Goto (from-trunk,51104,1)
-- Executing [51104@from-trunk:1] Set("SIP/trunk503in-0000010b", "__FROM_DID=51104") in new stack
-- Executing [51104@from-trunk:2] Gosub("SIP/trunk503in-0000010b", "app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/trunk503in-0000010b", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/trunk503in-0000010b", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/trunk503in-0000010b", "") in new stack
-- Executing [51104@from-trunk:3] Gosub("SIP/trunk503in-0000010b", "cidlookup,cidlookup_1,1") in new stack
-- Executing [cidlookup_1@cidlookup:1] GotoIf("SIP/trunk503in-0000010b", "1?cidlookup,cidlookup_return,1") in new stack
-- Goto (cidlookup,cidlookup_return,1)
-- Executing [cidlookup_return@cidlookup:1] ExecIf("SIP/trunk503in-0000010b", "0?Set(CALLERID(name)=)") in new stack
-- Executing [cidlookup_return@cidlookup:2] Return("SIP/trunk503in-0000010b", "") in new stack
-- Executing [51104@from-trunk:4] ExecIf("SIP/trunk503in-0000010b", "0 ?Set(CALLERID(name)=3145152244)") in new stack
-- Executing [51104@from-trunk:5] Set("SIP/trunk503in-0000010b", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:6] Set("SIP/trunk503in-0000010b", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:7] Goto("SIP/trunk503in-0000010b", "app-blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1] NoOp("SIP/trunk503in-0000010b", "Blackhole Dest: Hangup") in new stack
-- Executing [hangup@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@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@edj.devjones.com SIP/2.0
From: "Haley, Scott" <sip:3145152244@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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
If you are not the intended recipient of this message (including attachments), or if you have received this message in error, immediately notify us and delete it and any attachments.
If you do not wish to receive any email messages from us, excluding administrative communications, please email this request to messages@edwardjones.com along with the email address you wish to unsubscribe.
For important additional information related to this email, visit www.edwardjones.com/US_email_disclosure. Edward D. Jones & Co., L.P. d/b/a Edward Jones, 12555 Manchester Road, St. Louis, MO 63131 © Edward Jones. All rights reserved. |
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paul.belanger at polyb... Guest
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Posted: Wed Feb 26, 2014 11:14 am Post subject: [asterisk-users] SIP 603 Declined error message |
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On Wed, Feb 26, 2014 at 8:10 AM, Haley,Scott A
<scott.haley@edwardjones.com> wrote:
Quote: | 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@edj.devjones.com SIP/2.0
From: "Haley, Scott"
<sip:3145152244@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@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@edj.devjones.com>;avaya-cm-alert-type=internal
History-Info: <sip:51104@edj.devjones.com>;index=1
History-Info: "51104" <sip:51104@edj.devjones.com>;index=1.1
Min-SE: 1200
P-Asserted-Identity: "Haley, Scott" <sip:3145152244@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@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@192.168.122.51:5060;transport=TCP>
Content-Length: 0
<------------>
-- Executing [51104@from-trunk-sip-trunk503out:1]
Set("SIP/trunk503in-0000010b", "GROUP()=OUT_1") in new stack
-- Executing [51104@from-trunk-sip-trunk503out:2]
Goto("SIP/trunk503in-0000010b", "from-trunk,51104,1") in new stack
-- Goto (from-trunk,51104,1)
-- Executing [51104@from-trunk:1] Set("SIP/trunk503in-0000010b",
"__FROM_DID=51104") in new stack
-- Executing [51104@from-trunk:2] Gosub("SIP/trunk503in-0000010b",
"app-blacklist-check,s,1") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/trunk503in-0000010b",
"0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/trunk503in-0000010b",
"CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("SIP/trunk503in-0000010b",
"") in new stack
-- Executing [51104@from-trunk:3] Gosub("SIP/trunk503in-0000010b",
"cidlookup,cidlookup_1,1") in new stack
-- Executing [cidlookup_1@cidlookup:1] GotoIf("SIP/trunk503in-0000010b",
"1?cidlookup,cidlookup_return,1") in new stack
-- Goto (cidlookup,cidlookup_return,1)
-- Executing [cidlookup_return@cidlookup:1]
ExecIf("SIP/trunk503in-0000010b", "0?Set(CALLERID(name)=)") in new stack
-- Executing [cidlookup_return@cidlookup:2]
Return("SIP/trunk503in-0000010b", "") in new stack
-- Executing [51104@from-trunk:4] ExecIf("SIP/trunk503in-0000010b", "0
?Set(CALLERID(name)=3145152244)") in new stack
-- Executing [51104@from-trunk:5] Set("SIP/trunk503in-0000010b",
"__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:6] Set("SIP/trunk503in-0000010b",
"CALLERPRES()=allowed_not_screened") in new stack
-- Executing [51104@from-trunk:7] Goto("SIP/trunk503in-0000010b",
"app-blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1] NoOp("SIP/trunk503in-0000010b",
"Blackhole Dest: Hangup") in new stack
-- Executing [hangup@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@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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@edj.devjones.com SIP/2.0
From: "Haley, Scott"
<sip:3145152244@edwardjones.com>;tag=8066eb6f589ce3124b652973b4b00
To: <sip:51104@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) ---
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You'll want to talk to the FreePBX guys, as you are just hanging up
the outbound call.
-- Executing [51104@from-trunk:7] Goto("SIP/trunk503in-0000010b",
"app-blackhole,hangup,1") in new stack
-- Goto (app-blackhole,hangup,1)
-- Executing [hangup@app-blackhole:1]
NoOp("SIP/trunk503in-0000010b", "Blackhole Dest: Hangup") in new stack
-- Executing [hangup@app-blackhole:2]
Hangup("SIP/trunk503in-0000010b", "") in new stack
--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger
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