<html><head></head><body style="word-wrap: break-word; -webkit-nbsp-mode: space; -webkit-line-break: after-white-space; ">Any possibility that the faulty gateways have changed their architecture to hide media relays behind NAT, and that you have rtp-auto-adjust disabled or not correctly configured on your SIP profile ?<div>You may see that in the logs.</div><div>Another way to check this is to look the IP advertised by the gateway in the SDP, and to see if your FS is trying to send traffic to that IP.</div><div><br></div><div><div>
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</div>
<br><div><div>Le 21/04/2011 à 03:16, Fraser Redmond a écrit :</div><br class="Apple-interchange-newline"><blockquote type="cite">But it's odd that it's only happening with some gateways, but other gateways get two-way audio fine. That's what really has me puzzled.<br><br>It's definitely not a problem at the extension side, as it happens with multiple phones, and multiple locations. (So not mic, firewall, router)<br>
<br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin <span dir="ltr"><<a href="mailto:sos@sokhapkin.dyndns.org">sos@sokhapkin.dyndns.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
No outgoing audio? Usually this happens if SIP ALG is enabled in your router.<br>
<div><div></div><div class="h5"><br>
On Wednesday 20 April 2011, Fraser Redmond wrote:<br>
> No, I tried turning off the firewall, and as I said in the OP it works with<br>
> one of our other gateways.<br>
><br>
> Mic works on that one gateway, and during the calls where the audio isn't<br>
> transmitted the mic-indicator goes up and down.<br>
><br>
> Cheers,<br>
> Fraser<br>
><br>
> On Wed, Apr 20, 2011 at 5:46 PM, Brian West <<a href="mailto:brian@freeswitch.org">brian@freeswitch.org</a>> wrote:<br>
> > firewall issue? mic doesn't work? can you get a pcap of all traffic?<br>
> ><br>
> > /b<br>
> ><br>
> > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:<br>
> ><br>
> > Thanks Brian, I'd appreciate you looking - I don't know what to look for<br>
> > in the sip traces (could be worth documenting some pointers in the<br>
> > wiki?)<br>
> ><br>
> > The sip trace is here:<br>
> > <a href="http://pastebin.freeswitch.org/16136" target="_blank">http://pastebin.freeswitch.org/16136</a><br>
> ><br>
> > I pressed enter a few times in the console before and after it connected<br>
> > to the extension, so about lines 490-600 is the relevant part.<br>
> ><br>
> > I also captured a pcap, in case that is of interest - let me know and<br>
> > I'll email it directly.<br>
> ><br>
> > Thanks,<br>
> > Fraser<br>
> ><br>
> ><br>
> ><br>
> > _______________________________________________<br>
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