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

Tue Feb 20 19:40:15 CST 2007

Originally, my question about this was going to be, "How would
Asterisk know when your IP address changes?"

But, seeing your externrefresh=10 line, I am guessing this makes
Asterisk re-resolve your DNS name every 10 (seconds / minutes)?

If so, that is very cool, and should also solve Jayson's problem. I
was not aware of this feature in Asterisk. Is it new as of 1.4, or
have you been using that in 1.2 as well?

Rusty

On 2/20/07, Dean Clark <dgclark at oldphoneco.com> wrote:
> Correct me if I'm wrong, but I have a dynamic host set up through Dyndns
> and have added the following to my sip.conf, it seems to have alleviated
> much of my sip-nat-firewall-no static IP headaches.
>
> nat=yes
> externhost=myhost.dyndns.com
> localnet=192.168.1.0/255.255.255.0
> externrefresh=10
>
>
> Dean Clark
>
>
>
>
>
> Rusty Dekema wrote:
> > 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
> >>>
> >>>
> >>> _______________________________________________
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> >>> VoIP at ckts.info
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> >>>
> >>>
> >> _______________________________________________
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> >>
> >>
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> >
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> >
>
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