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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