[Asterisk-Users] RE: SIP Registration Timeout, No FW

[Fredrik von Kantzow]

Alright,

This is the output now:

Sep 27 19:49:02 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call 6b8b4567327b23c6643c986966334873 at 10.0.0.2 for seqno 107
(Critical Request)
Sep 27 19:49:16 NOTICE[98311]: chan_sip.c:4035 sip_reg_timeout: Registration
for '8703040 at sipgate.de' timed out, trying again
sunset*CLI> sip debug
SIP Debugging Enabled
Retransmitting #3 (no NAT):
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 216.110.117.54:5060;branch=z9hG4bK21ddfcb0
From: <sip:8703040 at sipgate.de>;tag=as226eb0dd
To: <sip:8703040 at sipgate.de>
Call-ID: 6b8b4567327b23c6643c986966334873 at 10.0.0.2
CSeq: 108 REGISTER
User-Agent: Asterisk PBX
Expires: 1800
Contact: <sip:1012 at 216.110.117.54>
Event: registration
Content-Length: 0

Settings:

[general]
context         = from-sip
port            = 5060
bindaddr        = 10.0.0.2
externip        = 216.110.117.54
srvlookup       = no
callerid        = Asterisk
language        = en
maxexpirey      = 3600
defaultexpirey  = 1800

register => SIP_ID:SIP_PW at sipgate.de/1012

[sipgate]
type=friend
host=sipgate.de
fromdomain=sipgate.de
secret=SIP_PW
username=SIP_ID
fromuser=SIPID
dtmfmode=info
canreinvite=yes
insecure=very

Still getting those timeouts :(
-Fredrik 

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com 
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of 
> Kai-Uwe Jensen
> Sent: Monday, September 27, 2004 7:07 PM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] RE: SIP Registration Timeout, No FW
> 
>  Nope, I'm sitting behind a router/firewall, with port 
> forwarding enabled.
>  Also using "externip = my.dyndns.hostname", automatically 
> updated by the  router.
> 
>  That should not be relevant, though, in this case. The 
> syntax for a SIP  register per the Wiki is
> 
>   register => user:secret:authuser@ ...
> 
>  You have it as "mynr:myuser:mypass@", or 
> "user:authuser:secret@". Also,  all sipgate needs is your 
> 7-digit SIP ID plus the corresponding password  (the one 
> assigned by sipgate).
> 
>  On one of the German VoIP forums there was talk about the 
> fact that in  the peer definition "fromdomain=sipgate.net" 
> *used* to work. However,  sipgate did change that and 
> apparently requires the fromdomain to be  sipgate.de.
> 
>  The other thing in the debug that strikes me is that your 
> local call IDs  are tied to the loopback interface 
> (127.0.0.1). I have no clue on whether  that works or not, 
> however I did include a "bindaddr" statement in my SIP  
> [general] context. Here's what it reads:
> 
>  [general]
>  disallow=all
>  allow    = ulaw
>  tos      = lowdelay
>  port     = 5060
>  bindaddr = 192.168.254.80 ; internal, non-natted IP address 
> to bind to  externip = my.dyndnsname.net  localnet = 
> 192.168.254.0/255.255.255.0  srvlookup = no  context  = 
> from-sip  callerid = Asterisk  language = en
>  maxexpirey     = 3600
>  defaultexpirey = 1800
> 
>  register => mynr:mypass at sipgate.de/mynr
> 
> 
> > Does your Asterisk server have a public IP?
> 
> > -Fredrik
> 
> > I'll only comment on sipgate, as I have it running without problems:
> > 
> > > register => mynr:myuser:mypass at sipgate.de/1012
> > 
> > Change this to
> > 
> >  register => mynr:mypass at sipgate.de/1012
> > 
> > (mypass is the SIP password, not your web login password).
> > 
> > Here's my sipgate context:
> > 
> >  [sipgate]
> >  type=friend
> >  host=sipgate.de
> >  fromdomain=sipgate.de
> >  secret=mypass
> >  username=mynr
> >  fromuser=mynr
> >  dtmfmode=info
> >  canreinvite=yes
> >  insecure=very
> > 
> > Note that fromdomain (sipgate.DE) and fromuser (mynr) differ from 
> > yours.
> > Also, note that in order to receive calls from sipgate.de, 
> you'll have 
> > to add the insecure=very if you're on a somewhat recent 
> release of *.
> 
> 
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