[Asterisk-Users] SIP registration

Fri Sep 19 00:28:47 MST 2003

No, it is not something you can fix by tweaking the configuration files, 
you should complain to the authors of the user agent.

Anyway, it is a minor problem and I guess that most implementations can
overcome it, but you should at least report it to the authors.

  Jan.

On 19-09 09:17, Sergio Serrano Revuelto wrote:
> Thanks, my phone has the next sip setting. Can you help me with correct
> parameters with the below sip.conf?
> 
> SIP Server Settings   
>  * Server Address:   (IP or FQDN) 
>  * Port:   
>  * Domain Name:   
>  * Send Registration Request:  (true or false)
>  
> Gateway Settings 
>  Dial Plan:   
>  Transport:  (UDP tor TCP )
>   
>   Phone Number:
>   CallerID Name: 
>   Port: 
>   AEC: (On or OFF)
>   User Name: 
>   Password: 
>  
> 
> 
> Thanks for all
> 
> 
> srsergio
> 
> 
> 
> 
> -----Mensaje original-----
> De: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com] En nombre de Jan Janak
> Enviado el: viernes, 19 de septiembre de 2003 8:59
> Para: asterisk-users at lists.digium.com
> Asunto: Re: [Asterisk-Users] SIP registration
> 
> 
> Hello,
> 
> I don't know if it is the problem, but the message below is
> syntactically invalid, there must be space between the name token in
> >From and To (704) and the URI, i.e. correct From should look like this:
> 
> 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
> > > 
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
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> > Asterisk-Users at lists.digium.com 
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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