Hi Warren,<br><br>Device A is behind NAT with regards to asterisk server. As far as localnet statement first I did configured localnet = <a href="http://130.8.2.0/255.255.255.0">130.8.2.0/255.255.255.0</a> as per local network, after that made a SIP call and the message I'm getting is listed below;<br>
<br>[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 <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a><br>
Packet timed out after 6400ms with no response<br>[Feb 2 11:14:52] WARNING[23868]: chan_sip.c:3651 retrans_pkt: Hanging up call OGEzODA0MzI4MWE1NzdiZDlkNmQ3NjYyMzJjYzYyOTY. - no reply to our critical packet (see <a href="https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions">https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions</a>).<br>
<br>So after setting to <a href="http://130.0.0.0/130.0.0.0">130.0.0.0/130.0.0.0</a> I wasn't getting the above warning message but facing one way audio.<br><br><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Date: Wed, 1 Feb 2012 14:38:01 -0600<br>
From: Warren Selby <<a href="mailto:wcselby@selbytech.com">wcselby@selbytech.com</a>><br>
Subject: Re: [asterisk-users] Getting one way audio even NAT is<br>
configured<br>
To: Asterisk Users Mailing List - Non-Commercial Discussion<br>
<<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
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<br>
On Wed, Feb 1, 2012 at 1:16 PM, Ahmed Munir <<a href="mailto:ahmedmunir007@gmail.com">ahmedmunir007@gmail.com</a>> wrote:<br>
<br>
> Hi all,<br>
><br>
> I'm getting one way audio when calling over the SIP trunk i.e. end device<br>
> B (remote end of SIP trunk) can hear device A (softphone registered with<br>
> Asterisk) but device A can't hear device B. Even though I configured same<br>
> NAT configurations on other servers and they are working good. The NAT<br>
> configuration is listed below;<br>
><br>
> localnet=<a href="http://130.0.0.0/130.0.0.0" target="_blank">130.0.0.0/130.0.0.0</a><br>
> externhost=12.131.12.13<br>
> externrefresh=10<br>
> fromdomain=<a href="http://test.localhost.com" target="_blank">test.localhost.com</a><br>
> nat=yes<br>
> qualify=yes<br>
> canreinvite=no<br>
><br>
><br>
> NAT on device end i.e. my softphone (extension) has already set to yes<br>
> with canreinvite=no but still unable to resolve this issue. SIP traces are<br>
> listed below;<br>
><br>
><br>
<snip><br>
<br>
<br>
><br>
> The Asterisk version I'm using is 1.8.5. Please assist me at earliest.<br>
><br>
<br>
Which device (A or B) is behind NAT with regards to your asterisk server?<br>
Is that the actual localnet= statement you're using, because to my<br>
understanding that is not the proper format to use (should be<br>
localnet=x.x.x.x/y.y.y.y where x.x.x.x is your actual local network, and<br>
y.y.y.y is your subnet for your local network).<br>
<br>
--<br>
Thanks,<br>
--Warren Selby, dCAP<br>
<a href="http://www.SelbyTech.com" target="_blank">http://www.SelbyTech.com</a> <<a href="http://www.selbytech.com" target="_blank">http://www.selbytech.com</a>><br>
</blockquote><div><br>--</div><div></div></div>Regards,<br><br>Ahmed Munir Chohan<br><br><br>