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 10:10:25 -0800 Message-ID: <49396E91.1060403@redfish-solutions.com> 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: Patrick McHardy Return-path: Received: from mail.redfish-solutions.com ([66.232.79.143]:41153 "EHLO mail.redfish-solutions.com" rhost-flags-OK-OK-OK-OK) by vger.kernel.org with ESMTP id S1754183AbYLESKb (ORCPT ); Fri, 5 Dec 2008 13:10:31 -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. > > What about the module options? pbx*CLI> sip show settings pbx*CLI> Global Settings: ---------------- SIP Port: 5060 Bindaddress: 0.0.0.0 Videosupport: No AutoCreatePeer: No Allow unknown access: Yes Allow subscriptions: Yes Allow overlap dialing: No Promsic. redir: No SIP domain support: Yes Call to non-local dom.: Yes URI user is phone no: No Our auth realm asterisk Realm. auth: No Always auth rejects: No Call limit peers only: No Direct RTP setup: No User Agent: Asterisk PBX MWI checking interval: 10 secs Reg. context: (not set) Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off IP ToS SIP: CS3 IP ToS RTP audio: EF IP ToS RTP video: AF41 T38 fax pt UDPTL: No RFC2833 Compensation: No SIP realtime: Disabled Global Signalling Settings: --------------------------- Codecs: 0x6 (gsm|ulaw) Codec Order: ulaw:20,gsm:20 T1 minimum: 100 Relax DTMF: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 0 (Disabled) RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: No Reg. min duration 60 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Default Settings: ----------------- Context: INVALID Nat: RFC3581 DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: (Defaults to English) MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk ---- pbx*CLI>