[asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Danny Nicholas
danny at debsinc.com
Mon Dec 22 08:40:46 CST 2008
This article might prove useful.
http://en.wikipedia.org/wiki/Iptables
telnet could provide the missing piece(s) of this puzzle for you.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of obitori junk
Sent: Friday, December 19, 2008 10:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem configuring Asterisk as client for
ekiga.net -- NAT problem
Danny:
I added
codecprobe=yes
did a restart and got the same error.
I changed it to "codecprobe=no" and got the same error. Is it possible
that the error is similar to this:
http://www.nabble.com/ekiga-registration-in-asterisk-td15816700.html
The reason I ask this is that my sip debug logs show:
From: "asterisk" <sip:asterisk at 10.1.1.40>;tag=as183adb5e
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f
Asterisk is putting itself out as "10.1.1.40" which obviously is my
internal LAN IP and not something that can be reached by ekiga.net.
I've used both stunaddr=stun.ekiga.net and externip=71.xxx.xxx.xxx (my
router's IP address).
I double checked my router. I'd not updated my IP address for the UDP
port forwarding since installing my new server (that is running
asterisk). But, even after making sure 10000-20000 is being forwarded
to my box, I don't see a single packet being logged by my firewall
(either accepted or blocked--I fiddled with logging to try both) from
ekiga.net's ip address. Also, junctn.net's sip connection is working no
problem. I'm stumped. :(
Any ideas out there?
Thanks for the help so far...Still plugging away...
Bud
On Fri, 2008-12-19 at 17:00 -0600, Danny Nicholas wrote:
> Check you codecprobe settings
> http://lists.digium.com/pipermail/asterisk-dev/2006-October/024101.html
>
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of obitori junk
> Sent: Friday, December 19, 2008 4:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Problem configuring Asterisk as client for
> ekiga.net -- NAT problem
>
> Danny:
>
> I restarted asterisk both times after making changes to the sip.conf in
> response to Eric's suggestions.
>
> I'll send you my original post with sip debug info. I ran sip debug
> again. The logged info is identical to what I was getting before I
> changed the codec allow/disallow settings...
>
> I think this is an Asterisk behind a NAT configuration problem, but if I
> could be wrong.
>
> Bud
>
>
> On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote:
> > Have you done a sip set debug, then sip reload? Do you have a range of
> > 10000-20000 open in your firewall? Asterisk will "poke out" through
5060
> > but has to get a random response back in the 10-20K range (you can
narrow
> > this)
> >
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of obitori
junk
> > Sent: Friday, December 19, 2008 4:28 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] Problem configuring Asterisk as client
for86.83.214.233
> > ekiga.net -- NAT problem
> >
> > I tried both "disallow all /// allow ulaw" and "allow all". Neither
> > worked. Ekiga.net is not accepting asterisk as a client. I keep
> > getting the "600 not acceptable error". What seems odd to me is that
> > the NAT column shows an "n" for both ekiga.net and jnctn.net, but I have
> > "nat=yes" for both. Thanks for the suggestion...I'm still open to
> > ideas.
> >
> > Regards,
> >
> > Bud Roth
> >
> > sip.conf contains: "allow all"
> > OUTPUT-->
> >
> > Name/username Host Dyn Nat ACL Port
> > Status
> > ekiga/budzhaus 86.64.162.35 N 5060 OK (102
> > ms)
> > 110 (Unspecified) D 0
> > UNKNOWN
> > 109 (Unspecified) D 0
> > UNKNOWN
> > 108 (Unspecified) D 0
> > UNKNOWN
> > 107 (Unspecified) D 0
> > UNKNOWN
> > 106 (Unspecified) D 0
> > UNKNOWN
> > 105 (Unspecified) D 0
> > UNKNOWN
> > 104 (Unspecified) D 0
> > UNKNOWN
> > 103 (Unspecified) D 0From: "asterisk"
<sip:asterisk at 10.1.1.40>;tag=as183adb5e
To: <sip:ekiga.net>;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f
> > UNKNOWN
> > 102/102 10.1.1.20 D 5061 OK (1
> > ms)
> > 101 (Unspecified) D 0
> > UNKNOWN
> > jnctn/obitori 66.227.100.20 N 5060 OK (22
> > ms)
> > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0
> > offline]
> > [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout: --
> > Registration for 'budzhaus at ekiga.net' timed out, trying again (Attempt
> > #2)
> > -- Got SIP response 606 "Not Acceptable" back from 86.64.162.35
> >
> > sip.conf contains:
> >
> > disallow all
> > allow ulaw
> >
> > OUTPUT-->
> > victoria*CLI> sip show peers
> > Name/username Host Dyn Nat ACL Port
> > Status
> > ekiga/budzhaus 86.64.162.35 N 5060 OK (97
> > ms)
> > 110 (Unspecified) D 0
> > UNKNOWN
> > 109 (Unspecified) D 0
> > UNKNOWN
> > 108 (Unspecified) D 0
> > UNKNOWN
> > 107 (Unspecified) D 0
> > UNKNOWN
> > 106 (Unspecified) D 0
> > UNKNOWN
> > 105 (Unspecified) D 0
> > UNKNOWN
> > 104 (Unspecified) D 0
> > UNKNOWN
> > 103 (Unspecified) D 0
> > UNKNOWN
> > 102/102 10.1.1.20 D 5061 OK (1
> > ms)
> > 101 (Unspecified) D 0
> > UNKNOWN
> > jnctn/obitori 66.227.100.20 N 5060 OK (23
> > ms)
> > 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0
> > offline]
> > victoria*CLI>
> >
> >
> > On Fri, 2008-12-19 at 15:18 -0600, Eric "ManxPower" Wieling wrote:
> > > obitori junk wrote:
> > > > I am experiencing a "606 not Acceptable" error trying to set up an
> > > > Asterisk server as an ekiga.net client. My server is behind a
> firewall
> > > > with NAT routing. I have googled this problem and read about
Asterisk
> > > > feeding its local ip address to ekiga.net. That seems to be my
> > > > problem.
> > >
> > > In my experience "Not Acceptable" errors happen because the two
> > > endpoints cannot agree on a codec. Try allowing all the codecs in
your
> > > softphone and in Asterisk sip.conf [general] do a disallow=all and an
> > > allow=ulaw. I suggest you do this in [general] when testing because
> > > it can sometimes be hard to make sure that a peer/friend/user entry is
> > > actually matching the incoming call. Once you get it working you can
> > > refine it.
> > >
> > > You could be having a NAT issue too, but I don't think it is related
to
> > > your 606 error.
> > >
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