No subject

From: 704 <sip:704 at AVANZADA7>;tag=230b0-e0

instead of this:

From: 704<sip:704 at AVANZADA7>;tag=230b0-e0

  Jan.

On 19-09 08:38, Sergio Serrano Revuelto wrote:
> I have the same problem,  
> 
> Asterisk debug is the next:
> 
> 
> REGISTER sip:AVANZADA7 SIP/2.0
> Call-ID: 45460-e1-c0a8145e at AVANZADA7
> From: 704<sip:704 at AVANZADA7>;tag=230b0-e0
> To: 704<sip:704 at AVANZADA7>
> CSeq: 101 REGISTER
> Via: SIP/2.0/UDP 192.168.0.154:5060
> Contact: sip:704 at 192.168.0.154:5060
> 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:704 at AVANZADA7>;tag=230b0-e0
> To: 704<sip:704 at AVANZADA7>;tag=as539680e1
> Call-ID: 45460-e1-c0a8145e at AVANZADA7
> CSeq: 101 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:704 at 192.168.0.207>
> Content-Length: 0
> 
> 
>  to 192.168.0.154:5060
> DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call
> '45460-e1-c0a8145e at AVANZADA7'
> 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:asterisk at 192.168.0.207>;tag=as6c232c12
> To: <sip:192.168.0.154>
> Contact: <sip:asterisk at 192.168.0.207>
> Call-ID: 4209223675d27d7c45ec94194860e7bb at 192.168.0.207
> 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: 4209223675d27d7c45ec94194860e7bb at 192.168.0.207
> From: asterisk<sip:asterisk at 192.168.0.207>;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: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] En nombre de Hielke
> Christian Braun
> Enviado el: jueves, 18 de septiembre de 2003 19:05
> Para: asterisk-users at lists.digium.com
> 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
> > 
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