[Asterisk-Users] Skinny protocol documentation

[Andre Normandin]

Thank you for the reply... I fixed it last night, although maybe someone can
explain to me why this occured in the first place, and how what I did fixed
it in the second place!!!!

Last night I decided to do some ethereal looking myself, and found the
following:

Indeed the RTP from the cisco is sending to the wrong IP address, but I have
no idea where it is getting the IP address from:

First, a little background on my setup:

(6 Routed IP addresses, x.y.z.24 - .31) from ISP goes into a linux box
acting as a firewall/internal nat, etc..

The Linux box is NOT set up as a router, I actually use iptables PREROUTING
to send ip addresses/protocol/ports to specific internal boxes to provide
services..

I have 2 internal networks (The firewall has 3 ethernet cards in it)

192.168.2.x is my DMZ where ALL my boxes that provide services to the
outside world reside

192.168.1.x is my internal network, and my asterisk is located on this
network.

The asterisk box itself has 3 X100P cards (3 analog lines in), and then I
have all SIP outbound (Using SPA-2000's for analog telephones, and
grandstream budgetone's for SIP telephones), plus 1 Cisco 7960 with the CM
image on it using chan_sccp.

The sip phones function fine, as do the X100 cards, and the phone lines (a
little echo problem with the grandstreams, but that's another 'help' message
<G>)

The asterisk server is at 192.168.1.223, as is the tftp server..

I configured the cisco phone with a hardcoded address of 192.168.1.203,
netmask 255.255.255.0..

Told it that the tftp is located on .223, the default GW is 192.168.1.254
(The linux box's interface on that subnet).

The cisco boots up ok, minus a cannot verify config info message (I only
have a XMLdefault.cnf.xml file on the tftp server), so I guess it's looking
for the other files (Sniffed that out last night so at least I know what
it's looking for now), but the lines appear correctly, can 'dial out', etc..
There is a problem calling the cisco, in that the cisco 'answers' the line
with a dial tone, and the calling extension has no idea that the cisco
picked up, and continues to ring the cisco.. But, I'll deal with that one
later..

Last night when I put ethereal on the network, the cisco was indeed sending
the RTP packets to x.y.z.26 (One of my 6 ACTUAL IP addresses).. I have NO
idea how it even found out that ip address. I did everything, modified the
DNS server so that .26 doesn't even appear in it, replacing it with the
internal .223 address etc.. Nothing helped..

Finally, I looked at my firewall rules on the firewall box, and I had this
one rule (At one time I had an http server on my asterisk box for testing
purposes, and forgot the rule was even in there)

iptables -t nat -A PREROUTING --in-interface ppp0 -d x.y.z.26 -p tcp --dport
80 -m state --state NEW -j DNAT --to-destination 1`192.168.1.223:80

I commented the rule out, and rebooted the cisco phone, and now I have audio
in both directions!!!

Does anyone have ANY idea why this would effect how the cisco phone decided
that the RTP should be sent to the external x.y.z.26 address in the first
place????


This is quite repeatable, I put the rule back in the firewall, reboot the
cisco, and again, I lose audio, and the cisso thinks it's sending it's rtp
packets to x.y.z.26 again.. Take it out, and it's back to working properly
again!!!!

 - Andre


-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Vic Cross
Sent: Wednesday, April 28, 2004 10:10 AM
To: asterisk-users at lists.digium.com
Subject: RE: [Asterisk-Users] Skinny protocol documentation


G'day,

I have used chan_sccp on 7960s with few problems.  I used nothing other
than the supplied sccp.conf as a basis for my configuration.  The phone
functions like Hold and Conference are probably not functional, but I
worked around this with Asterisk's own alternatives.

On Tue, 27 Apr 2004, Andre Normandin wrote:

> I cannot hear the conversation from the cisco to the other end.. I can
hear
> the other end on the cisco, but no audio the other direction..

I'd suggest running this down as a general one-way audio problem, rather
than something to do with SCCP in particular.  There's plenty of
information in the archives to help, but it usually involves a friendly
network analyser (ethereal).

One potential problem though: if your * has multiple network interfaces
(or alias IP addresses), you may be being bitten by the lack of a
"bindaddr" configuration parameter.  In one installation I had to patch
chan_sccp to do this, as the SDP data during call setup was indicating the
wrong IP address for one group of phones that were on a network that could
not route to the address chan_sccp chose.  Result: one-way audio...

Cheers,
Vic Cross
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