[VoIP] SIP/firewall problem - the saga

John Novack jnovack at stromberg-carlson.org
Sun Nov 12 16:37:12 CST 2006


The scenario here is, behind one router, an Asterisk box with a static 
private IP on the lan servicing an ATA pr SIP phone behind another 
router, also with a static private IP on another LAN
Many folks on the Asterisk users list have problems, especially one way 
audio.
Some time back Kirt and I attempted to get such a connection working, 
and found no solution, though I am sure there must be one.
Most of the examples don't fit, either with one device or another not 
behind a router, or in a router DMZ..

Beyond that I can't say.
That is your area of expertise

John Novack


john jones wrote:
> Are you using static NAT or dynamic NAT?  Typically, dynamic NAT is 
> the default and if the device on the inside does not continue to use 
> the connection, the translation is torn down after a relatively short 
> time.  Also, the NAT translation is usually specific to an individual 
> peer and can't be used by another external partner.
>
> John
>
> ----- Original Message ----
> From: John Novack <jnovack at stromberg-carlson.org>
> To: Voice Over IP Tandem for Analog Switches <voip at ckts.info>
> Sent: Sunday, November 12, 2006 3:31:07 PM
> Subject: Re: [VoIP] SIP/firewall problem - the saga
>
> In a situation where both ends are through routers with NAT, it is more
> problematical., and many have problems. One issue seems to be that NAT
> is available in several flavors, none documented with the equipment.
>
> For a SIP phone or ATA within one's own LAN it mostly just works.
>
>
> John Novack
>
>
> Andrew Green wrote:
> > In a Sipura the only configuration options you should need to set are,
> > "Display Name" (what you want the caller-id name to come up as),
> > "Proxy" (the ip/dns of the asterisk box), "Register = Yes", "User ID"
> > (the sip user in asterisk), and the Password (the secret in asterisk).
> > I never had to set any other options to get mine to work.
> >
> > Here is my asterisk sip context for my sipura-1001 (should also work
> > for a 2000):
> >
> > [spa-1001] ;you should set this to the same as the username
> > type=friend
> > host=dynamic
> > username=spa-1001 ;this is the User ID
> > secret=spa ;this is the password
> > context=default
> >
> > Also about the Draytek issue, try changing the registration timout to
> > about a minute so it will re-register every minute. That has helped a
> > friend's spa-1001 that was having the same problem.
> >
> >
> > Hope that helps a bit,
> > Andrew
> >
> >
> >
> > On 11/12/06, Ian Jolly <ian at uax.org.uk> wrote:
> >  
> >> Hi All
> >> from the other side of the Pond!
> >>
> >> Glad to know that we aren't the only ones having problems with SIP.
> >>
> >> When I do a 'sip show peers' mine are all 5060.  However when I do 
> an 'iax2
> >> show peers' I've got one  on Port 37657, it is on my own network -
> >> AT323/AT323      81.174.170.48   
> (D)  255.255.255.255  37657         OK (833
> >> ms)
> >>
> >>
> >> We've had success with one single CNET number off my *box working 
> with SIP
> >> to a Fritz AVM router with two VoIP ports on it.  Works OK bothways
> >> (although we haven't sorted out the dyndns set yet).
> >>
> >> However if we transfer exactly the same settings to a Sipura 2000 
> at the
> >> same location - it will not register and I can't see it when I do a 
> 'sip
> >> show peers'.
> >>
> >> Anyone any ideas?  Anyone got a working configuration for a Sipura 
> 2000 end?
> >>
> >> I also get this message every few seconds for a period of about 30 
> second
> >> occasionally - could this be why the above link became 'unable to 
> register'
> >> for 24 hours then just came back on without us doing anything at 
> either end?
> >>
> >> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:35:53 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:35:57 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:01 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:05 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:09 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:13 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:17 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >> Nov 12 19:36:21 WARNING[2485]: chan_sip.c:6393 register_verify: 
> Failed to
> >> parse contact info
> >>
> >>
> >>
> >> I am hosting another ATA - a Draytek 2600V with two VoIP ports one 
> of which
> >> is on CNET. We've got the SIP working OK bothways but every so 
> often  I get
> >> the following message at the CLI>-
> >>
> >> Nov 12 19:35:35 NOTICE[2485]: chan_sip.c:11328 sip_poke_noanswer: Peer
> >> 'jgriffiths' is now UNREACHABLE!  Last qualify: 226
> >> Nov 12 19:35:45 NOTICE[2485]: chan_sip.c:9689 
> handle_response_peerpoke: Peer
> >> 'jgriffiths' is now REACHABLE! (214ms / 2000ms)
> >>
> >> The Draytek becomes 'unreachable for ten seconds at a time every few
> >> minutes. During this period you can't dial into the number.  However it
> >> doesn't interrupt an existing call.  Anyone any idea what is 
> causing this
> >> 'unreachable/reachable' situation?  Incidentally the 
> 'chan_sip.c:11328' and
> >> 'sip.c:9689'  are always the same in the error messages - are they SIP
> >> ports?
> >>
> >> Ian Jolly
> >>
> >> ----- Original Message -----
> >> From: "Jayson Smith" <ratguy at bellsouth.net>
> >> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> >> Sent: Saturday, November 11, 2006 6:33 PM
> >> Subject: Re: [VoIP] SIP/firewall problem - the saga
> >>
> >>
> >>    
> >>> Hi,
> >>> HMMM, sound familiar?  Of course, in my situation, I'm appearing to be
> >>> accepting IAX2 connections on a random high port rather than 4569 
> (is that
> >>> the right port or am I losing my mind too?).  Yeah, some NAT 
> systems do
> >>> really strange things!
> >>> Jayson.
> >>>
> >>> ----- Original Message -----
> >>> From: "Chad Perkins" <chad at maine.maine.edu>
> >>> To: "Voice Over IP Tandem for Analog Switches" <voip at ckts.info>
> >>> Sent: Saturday, November 11, 2006 12:33 PM
> >>> Subject: Re: [VoIP] SIP/firewall problem - the saga
> >>>
> >>>
> >>>      
> >>>>> I hate SIP.
> >>>>> I don't know if this will help, but, when I do a "sip show peers," I
> >>>>> get some really odd ports for Dennis and for Jim:
> >>>>>
> >>>>>          
> >> _______________________________________________
> >> VoIP mailing list
> >> VoIP at ckts.info
> >> http://lists.ckts.info/mailman/listinfo/voip
> >> Project Web Page: http://www.ckts.info/
> >>
> >>
> >>    
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
> >
> >
> >  
> _______________________________________________
> VoIP mailing list
> VoIP at ckts.info
> http://lists.ckts.info/mailman/listinfo/voip
> Project Web Page: http://www.ckts.info/
>
>


More information about the VoIP mailing list