[asterisk-users] Call declined

Tue Nov 10 15:08:54 CST 2009

Thanks !!
it works

2009/11/10 Michael Wyres <mwyres at cdm.com.au>

>  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 --->
> 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
>
> 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091110/6a591e81/attachment-0001.htm 