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 19:08:09 +0100 Message-ID: <49396E09.9070607@trash.net> References: <4935B885.8030107@redfish-solutions.com> <493954ED.9070002@trash.net> <49396C4D.7020400@redfish-solutions.com> <49396DDD.9080308@trash.net> 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]:61447 "EHLO stinky.trash.net" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1753555AbYLESIM (ORCPT ); Fri, 5 Dec 2008 13:08:12 -0500 In-Reply-To: <49396DDD.9080308@trash.net> Sender: netfilter-devel-owner@vger.kernel.org List-ID: Patrick McHardy wrote: > Philip Prindeville wrote: >> Patrick McHardy wrote: >>> 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, ...)? >>> >> >> For the PSTN's switch: >> >> CanReinvite : Yes > > I vaguely recall some problem in the implementation of this feature, > something with missing bridging of the RTP streams that was still > necessary under some circumstances. Might be worth to try turning > it off. Actually I think it was directrtpsetup. Still worth trying though. > What about the module options?