[asterisk-users] Some give 603 Declined

Thu Oct 14 17:46:38 CDT 2010

Here is the sip log

ns*CLI> sip set debug peer hkbn2b
SIP Debugging Enabled for IP: 203.80.89.139:5060
[Oct 15 06:35:19] NOTICE[2462]: chan_sip.c:18334 handle_response_register:
Outbound Registration: Expiry for sip.voipuser.org is 120 sec (Scheduling
reregistration in 105 s)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [8935944101 at DLPN_DP1:1] Dial("SIP/6100-00000006",
"SIP/35944101 at hkbn2b,,r") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Audio is at 113.253.226.153 port 10650
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.80.89.139:5060:
INVITE sip:35944101 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as12eb85f9
To: <sip:35944101 at s2hkbntel.net:5060>
Contact: <sip:35944101hk at 113.253.226.153 <sip%3A35944101hk at 113.253.226.153>>
Call-ID: 3f603bea2560e9b836ea250932486935 at s2hkbntel.net
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.12
Date: Thu, 14 Oct 2010 22:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 241

v=0
o=root 316173620 316173620 IN IP4 113.253.226.153
s=Asterisk PBX 1.6.2.12
c=IN IP4 113.253.226.153
t=0 0
m=audio 10650 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called 35944101 at hkbn2b

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: <sip:35944101 at s2hkbntel.net:5060>
f: "cklee at mobile" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as12eb85f9
i: 3f603bea2560e9b836ea250932486935 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK1880eaca
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


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

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 487 Request Terminated
t: <sip:35944101 at s2hkbntel.net:5060>;tag=781480306
f: "cklee at mobile" <sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as12eb85f9
i: 3f603bea2560e9b836ea250932486935 at s2hkbntel.net
CSeq: 102 INVITE
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK1880eaca
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 203.80.89.139:5060:
ACK sip:35944101 at s2hkbntel.net:5060 SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK1880eaca;rport
Max-Forwards: 70
From: "cklee at mobile"
<sip:35944101hk at s2hkbntel.net<sip%3A35944101hk at s2hkbntel.net>
>;tag=as12eb85f9
To: <sip:35944101 at s2hkbntel.net:5060>;tag=781480306
Contact: <sip:35944101hk at 113.253.226.153 <sip%3A35944101hk at 113.253.226.153>>
Call-ID: 3f603bea2560e9b836ea250932486935 at s2hkbntel.net
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.12
Content-Length: 0


---
Scheduling destruction of SIP dialog '
3f603bea2560e9b836ea250932486935 at s2hkbntel.net' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [8935944101 at DLPN_DP1:2] Hangup("SIP/6100-00000006", "") in
new stack
  == Spawn extension (DLPN_DP1, 8935944101, 2) exited non-zero on
'SIP/6100-00000006'
[Oct 15 06:35:23] NOTICE[2462]: chan_sip.c:11601 sip_reregister:    --
Re-registration for  8887109919 at sip.pennytel.com
Reliably Transmitting (NAT) to 203.80.89.139:5060:
OPTIONS sip:s2hkbntel.net SIP/2.0
Via: SIP/2.0/UDP 113.253.226.153:5060;branch=z9hG4bK703ea06a;rport
Max-Forwards: 70
From: "asterisk"
<sip:asterisk at sip.etransmed.net<sip%3Aasterisk at sip.etransmed.net>
>;tag=as1d0ccbd8
To: <sip:s2hkbntel.net>
Contact: <sip:asterisk at 113.253.226.153 <sip%3Aasterisk at 113.253.226.153>>
Call-ID: 67f6129e02db3377276c62f209913543 at sip.etransmed.net
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.12
Date: Thu, 14 Oct 2010 22:35:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 100 Trying
t: <sip:s2hkbntel.net>
f: "asterisk" <sip:asterisk at sip.etransmed.net<sip%3Aasterisk at sip.etransmed.net>
>;tag=as1d0ccbd8
i: 67f6129e02db3377276c62f209913543 at sip.etransmed.net
CSeq: 102 OPTIONS
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK703ea06a
Server: MCS5x00_3.0
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


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

<--- SIP read from UDP:203.80.89.139:5060 --->
SIP/2.0 404 Not Found
t: <sip:s2hkbntel.net>;tag=820879923
f: "asterisk" <sip:asterisk at sip.etransmed.net<sip%3Aasterisk at sip.etransmed.net>
>;tag=as1d0ccbd8
i: 67f6129e02db3377276c62f209913543 at sip.etransmed.net
CSeq: 102 OPTIONS
v: SIP/2.0/UDP 113.253.226.153:5060
;received=113.253.226.174;rport;branch=z9hG4bK703ea06a
k: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec
l: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '
67f6129e02db3377276c62f209913543 at sip.etransmed.net' Method: OPTIONS


On Thu, Oct 14, 2010 at 7:55 AM, Paul Belanger <paul.belanger at polybeacon.com
> wrote:

> On Wed, Oct 13, 2010 at 6:48 PM, asterisk asterisk <asterisk at ck-lee.com>
> wrote:
> > Appreciate if help or direction can be provided.
> >
> 21.6.2 603 Decline
>
>   The callee's machine was successfully contacted but the user
>   explicitly does not wish to or cannot participate.  The response MAY
>   indicate a better time to call in the Retry-After header field.  This
>   status response is returned only if the client knows that no other
>   end point will answer the request.
>
> http://www.ietf.org/rfc/rfc3261.txt
>
> Collect a SIP trace and see if a reason is supplied.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101015/03994122/attachment.htm