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Hello,
I haven't understood exactly what you tried? Was it use of
force_send_socket(...) in onreply_route and didn't work? If yes, what
version are you runing, there was a fix for such case some time ago:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=9dbf735878acc87bab1a
Hello,
I don't know if iptel.org has TLS enabled, so I am replying just to say
that you can test with voipuser.org if you want -- another free voip
service. There is running latest kamailio and TLS is enabled.
Cheers,
Daniel
On 10/10/11 7:29 PM, Lello wrote:
Hi,
I'm working with a softpho
Hello,
On 10/10/11 6:43 AM, Austin Einter wrote:
If the SIP UA is not behind the NAT, and Kamailio insersts
a=nortpproxy:yes in SDP body, can it cause one sided audio.
The a= line should be added only when force_rtp_proxy() is used, which
should happen when is a call that involves NAT, unless y
Hi Daniel..
I'm using version 3.1.0. I looked at the change but its done only for
forward_reply.. How about relay_reply()?
Thanks
Jijo
On Thu, Oct 13, 2011 at 3:41 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> I haven't understood exactly what you tried? Was it use of
> force_send_socket(.
