[asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Danny Nicholas
danny at debsinc.com
Fri Dec 19 17:00:59 CST 2008
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 for
> 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 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 (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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>
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