[Asterisk-Users] SIP registration

Thu Sep 18 23:38:29 MST 2003

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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