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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of giancarlo lombardo Sent: Tuesday, 10 November 2009 09:03 To: asterisk-users@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:12...@192.168.1.100<mailto:sip%3a12...@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:gia...@192.168.1.116:14862<http://sip:gia...@192.168.1.116:14862>> To: "12345"<sip:12...@192.168.1.100<mailto:sip%3a12...@192.168.1.100>> From: "gianca"<sip:gia...@192.168.1.100<mailto:sip%3agia...@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:gia...@192.168.1.100<mailto:sip%3agia...@192.168.1.100>>;tag=db428348 To: "12345"<sip:12...@192.168.1.100<mailto:sip%3a12...@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:12...@192.168.1.100<mailto:sip%3a12...@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:12...@192.168.1.100<mailto:sip%3a12...@192.168.1.100>>;tag=as29d2b71c From: "gianca"<sip:gia...@192.168.1.100<mailto:sip%3agia...@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:12...@192.168.1.100<mailto:sip%3a12...@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:gia...@192.168.1.116:14862<http://sip:gia...@192.168.1.116:14862>> To: "12345"<sip:12...@192.168.1.100<mailto:sip%3a12...@192.168.1.100>> From: "gianca"<sip:gia...@192.168.1.100<mailto:sip%3agia...@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:12...@192.168.1.100<mailto:sip%3a12...@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:gia...@192.168.1.116:14862<http://sip:gia...@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:gia...@192.168.1.100<mailto:sip%3agia...@192.168.1.100>>;tag=db428348 To: "12345"<sip:12...@192.168.1.100<mailto:sip%3a12...@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:12...@192.168.1.100<mailto:sip%3a12...@192.168.1.100>> Content-Length: 0 <------------> -- Executing [12...@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:gia...@192.168.1.100<mailto:sip%3agia...@192.168.1.100>>;tag=db428348 To: "12345"<sip:12...@192.168.1.100<mailto:sip%3a12...@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:12...@192.168.1.100<mailto:sip%3a12...@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:12...@192.168.1.100<mailto:sip%3a12...@192.168.1.100>>;tag=as12cbf532 From: "gianca"<sip:gia...@192.168.1.100<mailto:sip%3agia...@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.
_______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
