[asterisk-users] Getting one way audio even NAT is configured

[Ahmed Munir]

Hi Warren,

Device A is behind NAT with regards to asterisk server. As far as localnet
statement first I did configured localnet = 130.8.2.0/255.255.255.0 as per
local network, after that made a SIP call and the message I'm getting is
listed below;

[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3622 retrans_pkt:
Retransmission timeout reached on transmission
OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. for seqno 1 (Critical
Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  2 11:14:52] WARNING[23868]: chan_sip.c:3651 retrans_pkt: Hanging up
call OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. - no reply to our
critical packet (see
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).

So after setting to 130.0.0.0/130.0.0.0 I wasn't getting the above warning
message but facing one way audio.

Date: Wed, 1 Feb 2012 14:38:01 -0600
> From: Warren Selby <wcselby at selbytech.com>
> Subject: Re: [asterisk-users] Getting one way audio even NAT is
>        configured
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>
> Message-ID:
>        <CAM_w8OJmX0nXfdU06p=-fpRAbZ2h7Tqr-mJMJNFnWeAvkjS+ZA at mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <ahmedmunir007 at gmail.com>
> wrote:
>
> > Hi all,
> >
> > I'm getting one way audio when calling over the SIP trunk i.e. end device
> > B (remote end of SIP trunk) can hear device A (softphone registered with
> > Asterisk) but device A can't hear device B. Even though I configured same
> > NAT configurations on other servers and they are working good. The NAT
> > configuration is listed below;
> >
> > localnet=130.0.0.0/130.0.0.0
> > externhost=12.131.12.13
> > externrefresh=10
> > fromdomain=test.localhost.com
> > nat=yes
> > qualify=yes
> > canreinvite=no
> >
> >
> > NAT on device end i.e. my softphone (extension) has already set to yes
> > with canreinvite=no  but still unable to resolve this issue. SIP traces
> are
> > listed below;
> >
> >
> <snip>
>
>
> >
> > The Asterisk version I'm using is 1.8.5. Please assist me at earliest.
> >
>
> Which device (A or B) is behind NAT with regards to your asterisk server?
> Is that the actual localnet= statement you're using, because to my
> understanding that is not the proper format to use (should be
> localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and
> y.y.y.y is your subnet for your local network).
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com <http://www.selbytech.com>
>

--
Regards,

Ahmed Munir Chohan
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