Thanks !! it works 2009/11/10 Michael Wyres <[email protected]>
> 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:* [email protected] [mailto: > [email protected]] *On Behalf Of *giancarlo lombardo > *Sent:* Tuesday, 10 November 2009 09:03 > > *To:* [email protected] > *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 ---> > INVITE sip:[email protected] <sip%[email protected]> 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:[email protected]:14862> > To: "12345"<sip:[email protected] <sip%[email protected]>> > From: "gianca"<sip:[email protected] <sip%[email protected]> > >;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:[email protected] <sip%[email protected]> > >;tag=db428348 > To: "12345"<sip:[email protected] <sip%[email protected]> > >;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:[email protected] <sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.116:14862 > ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport > To: "12345"<sip:[email protected] <sip%[email protected]> > >;tag=as29d2b71c > From: "gianca"<sip:[email protected] <sip%[email protected]> > >;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:[email protected] <sip%[email protected]> 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:[email protected]:14862> > To: "12345"<sip:[email protected] <sip%[email protected]>> > From: "gianca"<sip:[email protected] <sip%[email protected]> > >;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:[email protected] <sip%[email protected]> > ",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:[email protected]: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:[email protected] <sip%[email protected]> > >;tag=db428348 > To: "12345"<sip:[email protected] <sip%[email protected]>> > Call-ID: NzdlY2Y0Y2Q4Mjg1MGM4MGZkNjg3ZTZjYTI1OTM0NWY. > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Contact: <sip:[email protected] <sip%[email protected]>> > 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 --->** > 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:[email protected] <sip%[email protected]> > >;tag=db428348 > To: "12345"<sip:[email protected] <sip%[email protected]> > >;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:[email protected] <sip%[email protected]> SIP/2.0 > Via: SIP/2.0/UDP 192.168.1.116:14862 > ;branch=z9hG4bK-d8754z-dd20ff3e0753755b-1---d8754z-;rport > To: "12345"<sip:[email protected] <sip%[email protected]> > >;tag=as12cbf532 > From: "gianca"<sip:[email protected] <sip%[email protected]> > >;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 > -- Giancarlo Lombardo
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