asterisk users - Dec 2004 - Asterisk to sip client behind Firewall/NAT-cancall but cannot receive calls ?

As far as I can remember I only opened sip and tftp ports for the phone.

For some reason (didn't look into it too much) the call stays with sip
and doesn't use RTP.

The problem you describe (the call doesn't even ring on the other side)
is something I had and was solved by upgrading the firmware.

Checkpoint's tracker explicitly said what connection attempts were
blocked and why.

Check your logs and see.

Keep in mind that sip isn't part of the ANY group of protocols.
You need to either add it to ANY (not recommended) or set an explicit
rule for it.
> -----Original Message-----
> From: Robert Rozman [mailto:rozman@fri.uni-lj.si]
> Sent: Tuesday, December 14, 2004 1:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk to sip client behind
Firewall/NAT-
> cancall but cannot receive calls ?
> 
> Hi,
> 
> I hope I won't bother too much if I ask you to provide some more info
> about
> your setup, particularly which ports are open and other things (like
how
> often does Grandstream register, do you use keep alive, etc...).
> 
> Having that information I could rule out settings and maybe start
> searching
> on Firewalls...
> 
> Thanks in advance,
> 
> regards,
> 
> Robert.
> 
> ----- Original Message -----
> From: "Shoval Tomer" <shoval@softov.co.il>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Sent: Tuesday, December 14, 2004 11:30 AM
> Subject: RE: [Asterisk-Users] Asterisk to sip client behind
> Firewall/NAT -cancall but cannot receive calls ?
> 
> 
> Check your FW-1 tracker and see if any sip packets are dropped during
> call initiation.
> I had this problem and it went away when I upgraded the BT's firmware
to
> the latest (16).
> 
> Beware, though, that people on the list claim that this firmware
breaks
> functionality of the message button and autoanswer.
> I haven't checked this yet, cause I can't afford to go back a
version.
> I prefer a phone that can call then a phone that can autoanswer...
> 
> 
> 
> > -----Original Message-----
> > From: Robert Rozman [mailto:rozman@fri.uni-lj.si]
> > Sent: Tuesday, December 14, 2004 11:24 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: [Asterisk-Users] Asterisk to sip client behind Firewall/NAT
-
> > cancall but cannot receive calls ?
> >
> > Hi,
> >
> > I have following setup:
> >
> > BT100 ---- Firewall/nat 1 (www.ipcop.org) ---- Internet
> ----Firewall/nat2
> > (Vigor) ---- Asterisk .
> >
> > I'd like to use BT100 as local extension to Asterisk. I've
done
simple
> > setup
> > and BT100 can call Asterisk and place outgoing calls. However I
cannot
> set
> > him to qualify, cause it is claimed as unreachable.
> >
> > I have port redirection at Firewall 1 (to 5060 and rtp 5004 to
> > grandstream)
> > and 5060 and rtp ports on Firewall2. But I guess I am missing
> something.
> > I've setup Asterisk to work behing nat and it works OK, on same
route
> and
> > same local network Iax phone is operating also ok in both ways.
> >
> > On grandstream I've setup public NAT adress, then keep alive to 10
> sec,
> > (tried also some other setups but didn't work).
> >
> > I'm so close to working state, so would kindly ask for any
guidance
> (to
> > save
> > my hair) :-)
> >
> > Also I'm missing some understanding of SIP in this story: on what
> ports on
> > Asterisk machine does Grandstream connect RTP ?  Do I have to
transfer
> > some
> > other ports on Firewall2 because of Grandstream RTP connection ? If
> > everything works on outgoing calls there must be some little
condition
> > that
> > gets changed when I try to reach grandstream - what could that be ?
> >
> > If anyone has working scenarion, please be so kind to help.
> >
> >
> > Regards,
> >
> > Rob.
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > --
> > This message has been scanned for viruses and
> > dangerous content by MailScanner, and is
> > believed to be clean.
> > MailScanner thanks transtec Computers for their support.
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> --
> This message has been scanned for viruses and
> dangerous content by MailScanner, and is
> believed to be clean.
> MailScanner thanks transtec Computers for their support.

On Tuesday 14 December 2004 15:19, Shoval Tomer wrote:
> As far as I can remember I only opened sip and tftp ports for the phone.
>
> For some reason (didn't look into it too much) the call stays with sip
> and doesn't use RTP.
>
SIP is what sets up the session (ie it does session handling)
RTP is the transport protocol that the audio uses.

If you're using SIP then you're using RTP eos.

Jon