[Asterisk-Users] no sound when connected

Sat Apr 17 08:27:52 MST 2004

I assume you've got masquerading working so other hosts inside this
network are ok ?

something like...

/usr/local/sbin/iptables -v -t nat -A POSTROUTING -i eth0 -o eth1 -j
MASQUERADE ?

Definitely sounds more network than asterisk.

How about trying to connect to IAXTEL (which uses IAX2 rather than SIP) to
an external box ?  IAX2 protocol not SIP.  IAX2 is easier to get through
NAT and firewalls than SIP as it's brilliantly designed to just use one
UDP port I think (4569).
You perhaps try other sip destinations incase it's something with FWD
specifically.

you'll also need to keep the connection open.

Here's an extract from my sip.conf as to how I connect to FWD.  It may
help.

I've also got allow=alaw allow=ulaw allow=gsm in the general section of my
sip.conf as recommended by voip-info.org

register=user:passw at fwd.pulver.com/2030

[fwd]
type=friend
accountcode=fwd
disallow=all
allow=alaw
allow=ulaw
allow=gsm
username=XXX
secret=XXX
host=fwd.pulver.com
qualify=10000

you may well need nat=yes or other options.  Also are you registering with
fwd ok?

you could also run 'sip debug' and 'sip show peers' 'sip show users' and
asterisk with -d and -vvvv options to try and see what is going on.
Ethereal is also a very useful application for debugging you might want
to try?

I would assume if you are reaching the correct extension then dtmf is ok.
You want the RFC /outband if you're using a lossy codec (ie not A/U law)

Maybe you could also trying to connect to FWD directly with the
grandstream ?

Anyone else any pointers on what he could try - someone must have this
setup ??  Good luck!  Have patience and experiment and you'll crack it :)
SIP+NAT do not make wonderful companions but it is possible for sure :)

Chris

On Sat, 17 Apr 2004, Vlok Stone wrote:

> On Sat, 2004-04-17 at 14:43, Chris Orme wrote:
> > Very much sounds like a firewall issue not allowing voice packets back in 
> > to you (for the received audio) or them not finding you somehow.
> > 
> > Think about how do you connect to the internet.  Perhaps 'it' (whatver
> > device it is doing firewalling/NAT) is configurable through its bios or a
> > web interface or by telnet or ssh.  Depends what 'it' is, but 'it' is
> > likely to be involved in the problem.
> > 
> > You didn't send info of your configuration as to which protocol IAX/SIP
> > you are using and how you are trying to connect so I can't give a
> > specific answer on how to help you.  Or I didn't read closely enough.
> SIP is the protocol. 
> > 
> > I guess it might be:
> > 
> > BT 102 -SIP-> Asterisk on local LAN w/PSTN access/zap cards -SIP??->
> > firewall/router -adsl?-> internet -SIP-> Asterisk (2)
> yes that's the basic layout. firewall is linux w/ 2 nics iptables are
> down. I am able to ping out from the asterisk server. So, it is
> forwarding. 
> iptables -L
> Chain INPUT (policy ACCEPT)
> target     prot opt source               destination
> 
> Chain FORWARD (policy ACCEPT)
> target     prot opt source               destination
> 
> Chain OUTPUT (policy ACCEPT)
> target     prot opt source               destination
> 
> still no sound returns from FWD. 
> I'm sure it's the firewall, but can't figure out what's getting denied. 
> 
> > 
> > But I don't know as you didn't say :-(
> > 
> > I know Asterisk went through time when things weren't easy with
> > Grandstream phones, I don't know what the current state of affairs are
> > and I guess it is all great now if it working now via your Zap.
> > 
> > www.voip-info.org or using google to search the archives of this list
> > might also help, especially if you search perhaps for the name of your
> > firewall or router and asterisk or something like one way audio?  This is
> > what I did when I started.
> > 
> > Also if you have available other SIP clients to try on your network and
> > some patience I'm certain this can be tracked down and sorted out.
> > 
> > It might even be something as simple as 'nat=yes' 'host=dynamic'. There
> > are lots of sample configs on www.voip-info.org as well as those supplied
> > by Asterisk to work through.  Slowly change options from the sample config
> > and with patience you get the hang of things :-)
> I have nat=yes and host=dynamic
> > 
> > Hope that helps a little.  Just trying to put something back for all those
> > that helped me.
> Thank You. You're help is very much appreciated. I hope I may also be of
> assistance soon to others. 
> 
> > Good luck.
> > 
> > Chris
> > 
> > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > 
> > > On Sat, 2004-04-17 at 14:01, Chris Orme wrote:
> > > > Hi Vlok,
> > > > 
> > > > When a call connects is the audio one way ?  Can the remote person hear
> > > > you but you can't hear them ?
> > > yes. 
> > > > 
> > > > Which way is the audio or is it silent in both directions ?
> > > > The echo test?  Is this FWDs echo test or the one running on your
> > > > asterisk box (as that is not outside you LAN is it) ?
> > > ouside
> > > > 
> > > > I'm thinking this could be a NAT or firewall issue ?
> > > me too. what would i look for. 
> > > > 
> > > > Maybe you could give more info or a diagram of the set up you have there
> > > > so I can have a think about it?
> > > > 
> > > 
> > > > Chris
> > > > 
> > > > On Sat, 17 Apr 2004, Vlok Stone wrote:
> > > > 
> > > > > I'm having a sound issue. I'm using BT100 (102). When I dial the echo
> > > > > test ( or anything for that matter) outside of my LAN there's no sound
> > > > > when it answers although I hear the ringing tones. Is this an RTP or
> > > > > codec issue. When I dial through Zap everything is fine. Thanx.
> > > > > 
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