[asterisk-users] Call declined

Mon Nov 9 16:03:07 CST 2009

Dear all,
I'm in basic setup of my network:

I try to do a call from a softphone to an other one but I got the error 603
Declined.

Below the
sip.conf:
*[gianca]
type=friend
username=gianca
secret=pwd_gianca
host=dynamic
context=tutorial*
*[giusy]
type=friend
username=giusy
secret=pwd_giusy
host=dynamic
context=tutorial*

 extension.conf:
*[tutorial]
exten => 1234,1,Dial(SIP,gianca)*
*exten => 12345,1,Dial(SIP,giusy*)

Below the output of SIP debug of IP caller (192.168.1.116) in asterisk

*dhcppc0*CLI>
<--- SIP read from 192.168.1.116:14862 --->
INVITE sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:gianca at 192.168.1.116:14862>
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>>
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 265*
*v=0
o=- 6 2 IN IP4 192.168.1.116
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.116
t=0 0
m=audio 5960 RTP/AVP 107 0 8 101
a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv*
*<------------->
--- (12 headers 11 lines) ---
Sending to 192.168.1.116 : 14862 (NAT)
Using INVITE request as basis request -
NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.*
*<--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;received=192.168.1.116;rport=14862
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>
>;tag=as29d2b71c
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
upported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="42ebb35e"
Content-Length: 0*

*<------------>
Scheduling destruction of SIP dialog
'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)
Found user 'gianca'
dhcppc0*CLI>
<--- SIP read from 192.168.1.116:14862 --->
ACK sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>
>;tag=as29d2b71c
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 1 ACK
Content-Length: 0*

*<------------->
--- (7 headers 0 lines) ---
dhcppc0*CLI>
<--- SIP read from 192.168.1.116:14862 --->
INVITE sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:gianca at 192.168.1.116:14862>
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>>
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
Proxy-Authorization: Digest
username="gianca",realm="asterisk",nonce="42ebb35e",uri="
sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>
",response="8d00b3e1b28ed2e40681a3a9ee410046",algorithm=MD5
User-Agent: X-Lite release 1103k stamp 53621
Content-Length: 265*
*v=0
o=- 6 2 IN IP4 192.168.1.116
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.116
t=0 0
m=audio 5960 RTP/AVP 107 0 8 101
a=alt:1 1 : 6O20T816 ayXoiBh+ 192.168.1.116 5960
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=sendrecv*
*<------------->
--- (13 headers 11 lines) ---
Sending to 192.168.1.116 : 14862 (NAT)
Using INVITE request as basis request -
NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
Found user 'gianca'
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.116:5960
Found unknown media description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.116:5960
Looking for 12345 in tutorial (domain 192.168.1.100)
list_route: hop: <sip:gianca at 192.168.1.116:14862>*
*<--- Transmitting (no NAT) to 192.168.1.116:14862 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>>
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>>
Content-Length: 0*

*<------------>
    -- Executing [12345 at tutorial:1] Dial("SIP/gianca-088b96e0", "SIP|giusy")
in new stack
  == Spawn extension (tutorial, 12345, 1) exited non-zero on
'SIP/gianca-088b96e0'
Scheduling destruction of SIP dialog
'NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.' in 32000 ms (Method: INVITE)*
*<--- Reliably Transmitting (no NAT) to 192.168.1.116:14862 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;received=192.168.1.116;rport=14862
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>
>;tag=as12cbf532
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0*

*<------------>
dhcppc0*CLI>
<--- SIP read from 192.168.1.116:14862 --->
ACK sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116:14862
;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport
To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>
>;tag=as12cbf532
From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100>
>;tag=db428348
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 2 ACK
Content-Length: 0
*
**

-- 
Giancarlo Lombardo
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