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 at
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 at
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 at
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
IMPORTANT NOTICE TO RECIPIENT
Computer viruses - It is your responsibility to scan this email and any
attachments for viruses and defects and rely on those scans as Communications
Design & Management Pty Limited (CDM) does not accept any liability for loss
or damage arising from receipt or use of this email or any attachments.
Confidentiality - This email and any attachments are intended for the named
recipient only and may contain personal information, be it confidential or
subject to privilege, none of which are lost or waived because this email may
have been sent to you in error. If you are not the named addressee please let
CDM know by return email, permanently delete it from your system and destroy all
copies and do not use or disclose the contents.
Copyright - This email is subject to copyright and no part of it maybe
reproduced in any manner without the written permission of the copyright owner.
Privacy - Within the jurisdiction of Australian law, personal information in
this email must be dealt with in compliance with the Australian Federal Privacy
Act 1988.
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20091110/d4ed7cb5/attachment.htm