[VoIP] Sip, stupid sip, is biting me!

[Rusty Dekema]

Ok, upon taking more than 10 seconds to think about it, I agree with
Shane that the reinvite setting probably has nothing to do with your
problem.

If I were you, I would try this, even though it isn't a permanent
solution. Find out what your dynamically allocated IP address is, and
then enter that in the externip line in sip.conf under [general]. Then
see if you get two-way audio.

If this works, we will need to find a way to deal with your dynamic
IP. First of all, what kind of Internet connection is this (cable,
dsl, etc), and who is the provider? For example, the Charter cable
Internet service at my father's house provides, in theory, a dynamic
IP address, but it only changes once or twice a year. If yours is
anything like this, you could hardcode the IP in sip.conf and change
it whenever your IP changes.

Rusty




On 2/20/07, Jayson Smith <ratguy at bellsouth.net> wrote:
> Hi,
> Rtpstart and rtpend are set to 10000 and 20000 respectively, and that's what
> I've got forwarded. Putting "canreinvite=no" and "nat=yes" in the "general"
> section of sip.conf didn't help. I'm assuming here that externip is for
> specifying my external ip? If so, that may be needed, and this presents a
> problem, since my IP is dynamic. I do have a domain name through No-ip which
> gets updated whenever my IP changes, but I don't assume there's a way to
> have Asterisk automatically determine what my external IP is based on that?
> Thanks for all your help.
> Jayson.
>
> ----- Original Message -----
> From: "Shane Young" <voiptandem at shaneyoung.com>
> To: <voip at ckts.info>
> Sent: Tuesday, February 20, 2007 4:50 PM
> Subject: Re: [VoIP] Sip, stupid sip, is biting me!
>
>
> > The canreinvite really only applies to a call coming in SIP and going
> > to something else SIP, such as a SIP phone.
> >
> > What I understand to be Jayson's problem is simply any call coming in
> > to his system (and not going to another devices) is having one-way
> > audio.
> >
> > Make sure the ports which are forwarded through the firewall match
> > what is in the rtp.conf.
> >
> > For example:
> > rtpstart=16384
> > rtpend=32766
> >
> > You should be forwarding ports 16384 through 32766.  If not, the
> > asterisk box and the far end might negotiate a port which is blocked
> > by the firewall.
> >
> > You can also use ethereal or wireshark to see what's going on.
> >
> > Addtionally, check out the functionality for externip in your sip.conf
> > and see if that is needed for your setup or not.
> >
> > --Shane
> >
> >
> >
> >
> >
> >
> >
> >
> > Quoting Rusty Dekema <rdekema at gmail.com>:
> >
> > > On 2/20/07, Jayson Smith <ratguy at bellsouth.net> wrote:
> > >> Hello,
> > >> Isn't that just for a peer definition in sip.conf though? The way I
> have it
> > >> set up, Sipnumber has been told to call 622 at bluegrasspals.com to reach
> me.
> > >> Currently I have that pointed at an extension that plays a rather long
> > >> message, then runs the echo test to test audio. But anyone can call
> > >> 622 at bluegrasspals.com. There's no way to say "canreinvite=no" or
> "nat=yes"
> > >> for any and all random strangers that might call in, is there?
> > >> Jayson.
> > >
> > > I am not positive this will work, but I would try putting
> > > "canreinvite=no" under the [general] heading in sip.conf. I think the
> > > setting will then take effect for anonymous callers.
> > >
> > > Rusty
> > > _______________________________________________
> > > VoIP mailing list
> > > VoIP at ckts.info
> > > http://lists.ckts.info/mailman/listinfo/voip
> > > Project Web Page: http://www.ckts.info/
> > >
> >
> > --Shane
> > +1-821-7311 CNET
> >
> >
> > _______________________________________________
> > VoIP mailing list
> > VoIP at ckts.info
> > http://lists.ckts.info/mailman/listinfo/voip
> > Project Web Page: http://www.ckts.info/
> >
>
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