From mboxrd@z Thu Jan 1 00:00:00 1970 From: Philip Prindeville Subject: Re: More nf_conntrack_sip questions Date: Fri, 05 Dec 2008 10:00:45 -0800 Message-ID: <49396C4D.7020400@redfish-solutions.com> References: <4935B885.8030107@redfish-solutions.com> <493954ED.9070002@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: Patrick McHardy Return-path: Received: from mail.redfish-solutions.com ([66.232.79.143]:44558 "EHLO mail.redfish-solutions.com" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1753693AbYLESAx (ORCPT ); Fri, 5 Dec 2008 13:00:53 -0500 In-Reply-To: <493954ED.9070002@trash.net> Sender: netfilter-devel-owner@vger.kernel.org List-ID: 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: pbx*CLI> sip show peer sip_proxy pbx*CLI> * Name : sip_proxy Secret : xxxx MD5Secret : Context : ctc-incoming Subscr.Cont. : Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Nat : RFC3581 ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : 66.232.80.9 Addr->IP : 66.232.80.9 Port 5060 Defaddr->IP : 0.0.0.0 Port 0 Def. Username: SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw:20,gsm:20) Auto-Framing: No Status : Unmonitored Useragent : Reg. Contact : pbx*CLI> For the phone: pbx*CLI> sip show peer office_1 pbx*CLI> * Name : office_1 Secret : MD5Secret : Context : redfish-internal Subscr.Cont. : Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : Mailbox : 112@redfish,xxxx VM Extension : voicemail LastMsgsSent : 0/0 Call limit : 0 Dynamic : Yes Callerid : "Redfish Solutions" <112> MaxCallBR : 384 kbps Expire : 1780 Insecure : no Nat : Always ACL : No T38 pt UDPTL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Video Support: No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.1.7 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: office_1 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw:20,gsm:20) Auto-Framing: No Status : OK (9 ms) Useragent : Linksys/SPA942-5.1.15(a) Reg. Contact : sip:office_1@192.168.1.7:5060 pbx*CLI>