asterisk users - Sep 2010 - one way audio for xlite clients behind NAT

I am having a one way audio issue with xlite clients behind NAT. They can
connect to the server and make calls but no audio is heard on the other
end.

my sip conf

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
canreinvite=no[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip[1001];Work
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=sip.domain.com
nat=no[1000];IPKall
type=peer
dtmfmode=rfc2833
context=sip
insecure=very
host=voiper.ipkall.com
nat=no


I pasted the log here -> http://pastie.org/1163238


I have tried connecting both of the clients to another sip
service(sip2sip.info) and did not have the same problems.


Any suggestions would be great.

Thanks,

Tom
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100916/8037954e/attachment.htm

On 09/16/2010 06:58 PM, Thomas Johnson wrote:
> I am having a one way audio issue with xlite clients behind NAT. They
> can connect to the server and make calls but no audio is heard on the
> other end.
>
> my sip conf
>
> [general]
> context=default
> bindport=5060
> bindaddr=0.0.0.0
> srvlookup=yes
> canreinvite=no
>
> [tomfmason]
> type=friend
> secret=secret
> callerid="Thomas Johnson"  <XXXX>
> host=dynamic
> nat=yes
> canreinvite=no
> disallow=all
> allow=gsm
> allow=ulaw
> allow=alaw
> qualify=yes
> context=sip
>
> [1001];Work
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=sip.domain.com  <http://sip.domain.com>
> nat=no
>
> [1000];IPKall
> type=peer
> dtmfmode=rfc2833
> context=sip
> insecure=very
> host=voiper.ipkall.com  <http://voiper.ipkall.com>
> nat=no
You seem to be using nat=no

shouldn't that be nat=yes?
>
>
>
> I pasted the log here ->  http://pastie.org/1163238
>
>
> I have tried connecting both of the clients to another sip
service(sip2sip.info  <http://sip2sip.info>) and did not have the same
problems.
>
>
> Any suggestions would be great.
>
> Thanks,
>
> Tom
>

I already have that covered

[tomfmason]
type=friend
secret=secret
callerid="Thomas Johnson" <XXXX>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
qualify=yes
context=sip

The server is not behind NAT only the client above is

On Thu, Sep 16, 2010 at 4:59 PM, Paul Belanger <paul.belanger at
polybeacon.com
> wrote:
> On Thu, Sep 16, 2010 at 5:50 PM, Thomas Johnson <tomfmason at
gmail.com>
> wrote:
> > Also, if I disable the firewall in my router I lose incoming audio and
> > outgoing audio works.
> >
> http://www.aocomputing.net/?p=3
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100916/cf5a0e0e/attachment.htm

On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson <tomfmason at gmail.com>
wrote:
> The server is not behind NAT only the client above is
>
Sounds like a phone (not asterisk) issue then, make sure you have
setup your NAT and port forwarding properly on the client side.

-- 
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com

If you are using linux firewall, try this, it was very usefull to me:


iptables -t nat -A PREROUTING -i ethx -p tcp --dport 5060 -j DNAT --to
ip_phoneiptables -t nat -A PREROUTING -i ethx -p udp --dport 5060 -j DNAT --to
iip_phoneiptables -A FORWARD -p TCP --dport 5060 -j ACCEPTiptables -A FORWARD -p
UDP --dport 5060 -j ACCEPT



Att,

 

Flavio Roberto Miranda

MSN:flaviormiranda at hotmail.com
Skype: flaviormiranda


> Date: Thu, 16 Sep 2010 18:45:38 -0400
> From: paul.belanger at polybeacon.com
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] one way audio for xlite clients behind NAT
> 
> On Thu, Sep 16, 2010 at 6:04 PM, Thomas Johnson <tomfmason at
gmail.com> wrote:
> > The server is not behind NAT only the client above is
> >
> Sounds like a phone (not asterisk) issue then, make sure you have
> setup your NAT and port forwarding properly on the client side.
> 
> -- 
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belanger at polybeacon.com | IRC: pabelanger (Freenode)
> blog.polybeacon.com
> 
> -- 
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
 		 	   		  
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20100916/ed9ea44c/attachment-0001.htm