I have the same problem, Asterisk debug is the next:
REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704<sip:[EMAIL PROTECTED]>;tag=230b0-e0 To: 704<sip:[EMAIL PROTECTED]> CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704<sip:[EMAIL PROTECTED]>;tag=230b0-e0 To: 704<sip:[EMAIL PROTECTED]>;tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: "asterisk" <sip:[EMAIL PROTECTED]>;tag=as6c232c12 To: <sip:192.168.0.154> Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisk<sip:[EMAIL PROTECTED]>;tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -----Mensaje original----- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: > Hi, > > I'm having problems letting a SIP endpoint register at Asterisk. > Here's the > debug output from Asterisk: > > > ... > > sip.conf: > > [general] > port=5060 > bindaddr=s.s.s.s > context=cxnet-in > tos=lowdelay > > [siptestphone] > type=friend > user=atrg613test > host=dynamic > defaultip=c.c.c.c > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
