[asterisk-users] Call declined

Mon Nov 9 17:13:17 CST 2009

Try:

[tutorial]
exten => 1234,1,Dial(SIP/gianca,10,t)
exten => 12345,1,Dial(SIP/giusy,10,t)

You want a "/" between SIP and the name of the phone, not an ",".

The "10" refers to the number of seconds you want the phone to ring.  The "t" allows the channel to be transferred after pickup - not strictly needed, but I tend to put it in in most instances as generally you'll want it.

For more information on the Dial application, see http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of giancarlo lombardo
Sent: Tuesday, 10 November 2009 09:03
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Call declined

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<http://192.168.1.116:14862> --->
INVITE sip:12345 at 192.168.1.100<mailto: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<http://sip:gianca@192.168.1.116:14862>>
To: "12345"<sip:12345 at 192.168.1.100<mailto:sip%3A12345 at 192.168.1.100>>
From: "gianca"<sip:gianca at 192.168.1.100<mailto: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<http://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<mailto:sip%3Agianca at 192.168.1.100>>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100<mailto: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<http://192.168.1.116:14862> --->
ACK sip:12345 at 192.168.1.100<mailto: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<mailto:sip%3A12345 at 192.168.1.100>>;tag=as29d2b71c
From: "gianca"<sip:gianca at 192.168.1.100<mailto: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<http://192.168.1.116:14862> --->
INVITE sip:12345 at 192.168.1.100<mailto: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<http://sip:gianca@192.168.1.116:14862>>
To: "12345"<sip:12345 at 192.168.1.100<mailto:sip%3A12345 at 192.168.1.100>>
From: "gianca"<sip:gianca at 192.168.1.100<mailto: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<mailto: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<http://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<http://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<http://sip:gianca@192.168.1.116:14862>>
<--- Transmitting (no NAT) to 192.168.1.116:14862<http://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<mailto:sip%3Agianca at 192.168.1.100>>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100<mailto: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<mailto: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<http://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<mailto:sip%3Agianca at 192.168.1.100>>;tag=db428348
To: "12345"<sip:12345 at 192.168.1.100<mailto: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<http://192.168.1.116:14862> --->
ACK sip:12345 at 192.168.1.100<mailto: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<mailto:sip%3A12345 at 192.168.1.100>>;tag=as12cbf532
From: "gianca"<sip:gianca at 192.168.1.100<mailto:sip%3Agianca at 192.168.1.100>>;tag=db428348
Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY.
CSeq: 2 ACK
Content-Length: 0

--
Giancarlo Lombardo
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