[Asterisk-Users] Double NAT issues with SIP and workaround (?)

[Steve]

I use Linux/iptables for a firewall
and it seems that in sip.conf
canreinvite=no has been magic for me!

It's fixed one-way audio problems for me in just about every case I have 
ever run across so far.

This seems to be an 'easy fix' but I understand from other posts it might 
be a waste of Internet bandwidth in many cases.


I'm still a newbie (3 weeks at it now)

Have got all kinds of neat things to work and work great except still 
cannot get a successful outbound call to work via a sip provider! :-)
and I have tried more than just one sip service provider.
that's another thread....


Take care!

Steve











On Mon, 6 Jun 2005, Julian J. M. wrote:

> Hello,
>
> I've been fighting one-way-audio issues with asterisk and SIP
> extensions for some time..., and  I want to share with you my findings
> ;)
>
> My setup:
>    * 1 ADSL router (Zyxel)
>    * 1 Asterisk box with private IP, and interesting ports forwarded to it.
>    * Several extensions, some local some remote
>
> The problem:
>    * External extensions behind double nat don't get audio when they
> initiate a call. But if the extension receives the call, there is no
> problem.
>
> The fix:
>    * Get a Linksys WRT54G, and setup the adsl router in bridge mode,
> giving the public IP address to the WRT.
>   * Setup port forwarding and QoS (optional)
>   * Enjoy VoIP ;)
>
>
> I don't know the exact reasons why this happens, but it works ;).
>
> Julian.
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