From mboxrd@z Thu Jan 1 00:00:00 1970 From: Patrick McHardy Subject: Re: More nf_conntrack_sip questions Date: Fri, 05 Dec 2008 17:21:01 +0100 Message-ID: <493954ED.9070002@trash.net> References: <4935B885.8030107@redfish-solutions.com> Mime-Version: 1.0 Content-Type: text/plain; charset=ISO-8859-15; format=flowed Content-Transfer-Encoding: 7bit Cc: netfilter-devel@vger.kernel.org To: Philip Prindeville Return-path: Received: from stinky.trash.net ([213.144.137.162]:59462 "EHLO stinky.trash.net" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1751869AbYLEQVE (ORCPT ); Fri, 5 Dec 2008 11:21:04 -0500 In-Reply-To: <4935B885.8030107@redfish-solutions.com> Sender: netfilter-devel-owner@vger.kernel.org List-ID: Philip Prindeville wrote: > I did a little investigation into my one-way voice issue, and noticed > that if I don't do voice-menus (i.e. where the Asterisk box itself > generates the first outbound INVITE, then passes-through the 2nd INVITE > once a handset picks up) then I get two-way voice (i.e. with sending the > call directly to the phone). (In this topology, my Asterisk box is also > my firewall/NATting router...) > > If I enable the voice menus in the inbound dialplan, however, it can > hear the voice menus, but not the called-party when they pick up their > phone (extension). > > So someone (either the SIP conntrack module on the Asterisk border > firewall or else the SBC at the ILEC) is failing to look into the 2nd > INVITE (i.e. we're not rewriting it properly as it goes by, or the SBC > is failing to see it). What module options are you using for the SIP helper and how is call setup in asterisk configured (directrtpsetup, canreinvite, ...)?