- FreeSWITCH-users - FreeSWITCH Mailing Lists

ReINVITE failure with ACK+SDP scenario
by BNML 02 Jul '13

02 Jul '13

Hi all, I am having trouble with a transfert failure in a special scenario which i wasn't able to find documentation or configuration option solving it. I have a PBX creating two outbound call, it want to bridge them on the FreeSWITCH. It is sending a first INVITE without SDP and it will send the SDP in the ACK. (This is not breaking RFC3261 but I admit rarely seen scenario. http://tools.ietf.org/html/rfc3261#section-13.2.1 : "The UAC MAY choose to add a message body to the INVITE.". Looking at this scenario there is, indeed, at least one interesting scenario where it can helps avoiding creating another useless Re-INVITE transaction, see below) For the second call it is sending a re-invite with SDP. FreeSWITCH will understand and forward (because of "bypass_media") the second Re-INVITE but does never forward the first Re-INVITE (The one without SDP in the INVITE but in the ACK) to carrier, so this will end up in one-way-audio. A little schema to help : PBX FREESWITCH COMMENT 1) INVITE (a-leg) [THERE IS NO SDP] 2) 200 OK (fs-a-leg)[FreeSWITCH answer with the SDP information before any media update occurs, but my PBX is now aware of media informations] 3) INVITE+SDP (b-leg) [This is b-leg updated media information with previous 200 OK (fs-a-leg)] 4) 200 OK (fs-b-leg)[My PBX updated media and has b-leg freeswitch media information that it will send in the ACK+SDP to finalize the both-legs Re-INVITE] 5) ACK (b-leg) [Acknoledgement of b-leg Re-INVITE transaction] 6) ACK+SDP (a-leg) [ *** problem *** it assume FreeSWITCH will update media information and forward to the carrier which it won't] The solution i imagine is FreeSWITCH SHOULD trigger a Re-INVITE either for the a-leg at INVITE (without SDP in "1") receiption (and update media information when receiving the ACK+SDP) _OR_ create a Re-INVITE transaction at "6" when receiving ACK+SDP. One way or another, I am supposing FreeSWITCH refuse to forward the INVITE without SDP at first maybe because of carrier compatibility ? If not, this seems the way to go. Has anyone faced such a scenario ? Am I rigth assuming a patch is needed ? If yes, any suggestions are welcome. Thanks you BNML

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Updated 1.4.beta tarball
by Ken Rice 02 Jul '13

02 Jul '13

Hey guys, Check it out http://files.freeswitch.org/freeswitch-1.4.0.beta4.tar.bz2 Its out on the CDN... If you are tracking the v1.4.beta branch in git, and have updated recently you¹re on track! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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Re: [Freeswitch-users] Webrtc video
by Phil Dunks 02 Jul '13

02 Jul '13

Hi Ken, Anthony Thanks for the info. Is a solution for this on the radar, or would you advise using a 3rd party MCU for webrtc video? We're only after the pass-through stream for the current floor - I tried this with x-lites and H.264 and it worked well. Thanks, Phil > From: Ken Rice <krice(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Webrtc video > Date: 28 June 2013 16:47:11 BST > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > > > The Video handling of WebRTC is fine. The Video handling in general of mod_conference is where you are prob seeing problems...FreeSWITCH doesn’t trans code video so mod_conference depends on Key frames, and for what I understand vp8 doesn’t have key frames so that might be causing your problem... We use a 3rd party MCU for demoing video conferencing as see via calling mcu.freeswitch.org > > > > Hi FS Users > > I made a simple web application using sipML5, which connects directly to 1.4.0+git~20130623T182400Z~2f08e40fce, and goes straight into a conference. > The audio works fine, but am experiencing problems with the video freezing or not being present. > I tried out https://webrtc.freeswitch.org/webrtc/portal.html > Enabled video, and dialed 3300 for the local 48K conference. > Seeing the same problem there. Audio fine. Video fine for first member, but goes wrong when second person joins. > Should this work for me, or is the webrtc conference video not quite ready yet? > Thanks, > Phil

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Do not strip X headers
by Mino Haluz 02 Jul '13

02 Jul '13

Hi, I cannot remember the variable you should set when you want FS to do not strip custom X- headers received on inbound leg. Anybody? Thanks, M

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re-invite to bridge
by Jimmy Chang 02 Jul '13

02 Jul '13

Hi, I'm new in FS. I don't understand why FS doesn't forward invite to a gateway. Here is the scenario. 1) A SIP client make a SIP call to FS 2) FS bridge the call to a gateway 3) the client send the invite again (disable video) 4) FS doesn't forward the invite to gateway Is it normal? Or do I need to configure any parameter? Thanks.

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FreeSWITCH Weekly News and Notes
by Michael Collins 02 Jul '13

02 Jul '13

Welcome to July! We've had lot's of great news<http://online.wsj.com/article/PR-CO-20130624-907435.html> lately with FreeSWITCH and now there's even more. Our friends at Yealink have reported on a very large VoIP installation in Spain that utilizes Yealink phones and FreeSWITCH servers. The complete writeup can be found here <http://www.yealink.com/News_info.aspx?NewsId=330&CateId=306>. We look forward to reporting on more success stories in the future. With the recent announcement <http://freeswitch.org/node/455> about FreeSWITCH 1.4 beta and WebRTC support there has been a lot of interest in just how to make this all work. On this week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_07_03> we will be taking a closer look at the code and showing you how to setup both sides of a WebRTC connection with FreeSWITCH. Please join usthis Wednesday at 1PM Eastern, 10AM Pacific for this interesting discussion. We also had an interesting discussion on last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_26>. Lorenzo Magani and Alexandr Dubovikov from the HOMER<http://www.sipcapture.org/> project gave us a tour of their new version 3.5 release. The HOMER SIP capture system is an excellent - and free - tool that can be put into production with any FreeSWITCH server. HOMER is an invaluable tool used for SIP diagnostics and troubleshooting and we highly recommend it for anyone using FreeSWITCH and SIP in a production environment. Have a great week and we'll talk to you on Wednesday. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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effective_caller_id_name & effective_caller_id_number - not working
by Lloyd Aloysius 02 Jul '13

02 Jul '13

Hi All, When sending calls out from gateway the caller id name and number not working. Please find the freeswitch cli log and sip invite trace. Any help is appreciated. *freeswitch cli* EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_name=Alcan Cable) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara(a)alcan.mydomain.ca SET [effective_caller_id_name]=[Alcan Cable] EXECUTE sofia/sipinterface_1/barbara(a)alcan.mydomain.caset(effective_caller_id_number=9052066900) 2013-07-01 19:23:21.676990 [DEBUG] mod_dptools.c:1373 sofia/sipinterface_1/ barbara(a)alcan.mydomain.ca SET [effective_caller_id_number]=[9052066900] *SIP Trace* INVITE sip:4164750401@alcan.mydomain.ca SIP/2.0 Via: SIP/2.0/UDP 192.168.240.105:5060;branch=z9hG4bK17bfd0ad4 Max-Forwards: 70 Content-Length: 285 To: 4164750401 <sip:4164750401@alcan.mydomain.ca> From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105 CSeq: 1894816474 INVITE Supported: timer Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Content-Type: application/sdp Contact: Barbara Ladner <sip:barbara@192.168.240.105:5060;transport=udp> Supported: replaces User-Agent: Aastra 9133i/1.4.3.1001 Brcm Callctrl/1.5.1.0 MxSF/v3.2.8.45 v=0 o=MxSIP 0 575681134 IN IP4 192.168.240.105 s=SIP Call c=IN IP4 192.168.240.105 t=0 0 m=audio 3000 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.240.105:5060 ;branch=z9hG4bK17bfd0ad4;received=99.247.64.220;rport=16511 From: Barbara Ladner <sip:barbara@alcan.mydomain.ca>;tag=02f4abe9bef3a7f To: 4164750401 <sip:4164750401@alcan.mydomain.ca> Call-ID: 43016333178f7b0882c3767aab886c1d(a)192.168.240.105 CSeq: 1894816474 INVITE User-Agent: FreeSWITCH-mod_sofia/1.5.3b+git~20130628T160527Z~a52a604fbb Content-Length: 0 * * Thanks Lloyd

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ZRTP security bugs
by Ben 01 Jul '13

01 Jul '13

Hi, Does the recent news about the flaws in the ZRTPCPP library (http://blog.azimuthsecurity.com/2013/06/attacking-crypto-phones-weaknesses- in.html) impact the Freeswitch implementation of ZRTP in any way? Appreciate this is fairly fresh news, so in a way it's more of an FYI! Thanks.

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No voicemail after attempted FusionPBX upgrade
by Phil Quesinberry 01 Jul '13

01 Jul '13

Just attempted an upgrade to FusionPBX and it appears to have messed up lua. I completely backed up the FS config (/usr/local/freeswitch/conf) as well as the FusionPBX files and database before attempting this but reverting to the old configuration doesn't fix the problem, despite doing a reloadxml/restarting FS, etc. I think FusionPBX may have made some changes to the scripts directory which I hadn't backed up. Just to be sure, I recompiled with 'make current' but that didn't help either. I did make a snapshot of the VM that FS is running on a day or two ago but I'd like to understand what happened. I now get the following errors in the console output when attempting to enter voicemail: EXECUTE sofia/internal/102@192.168.1.6:5060 set(voicemail_action=check) 2013-07-01 15:10:24.774751 [DEBUG] mod_dptools.c:1373 sofia/internal/102@192.168.1.6:5060 SET [voicemail_action]=[check] EXECUTE sofia/internal/102@192.168.1.6:5060 set(voicemail_profile=default) 2013-07-01 15:10:24.774751 [DEBUG] mod_dptools.c:1373 sofia/internal/102@192.168.1.6:5060 SET [voicemail_profile]=[default] EXECUTE sofia/internal/102@192.168.1.6:5060 lua(app.lua voicemail) 2013-07-01 15:10:24.774751 [ERR] mod_lua.cpp:198 cannot open /usr/local/freeswitch/scripts/resources/functions/file_exists.lua: No such file or directory stack traceback: [C]: in function 'dofile' /usr/local/freeswitch/scripts/resources/config.lua:20: in main chunk [C]: in function 'dofile' /usr/local/freeswitch/scripts/app.lua:28: in main chunk 2013-07-01 15:10:24.774751 [DEBUG] switch_cpp.cpp:1050 sofia/internal/102@192.168.1.6:5060 destroy/unlink session from object 2013-07-01 15:10:24.774751 [NOTICE] switch_core_state_machine.c:262 sofia/internal/102@192.168.1.6:5060 has executed the last dialplan instruction, hanging up. 2013-07-01 15:10:24.774751 [NOTICE] switch_core_state_machine.c:264 Hangup sofia/internal/102@192.168.1.6:5060 [CS_EXECUTE] [NORMAL_CLEARING] Doing a sofia loglevel all 9 or SLA debug doesn't really seem to generate any additional useful info pertaining to this problem, at least not that I can see. Any ideas on what might have happened here? FusionPBX comes in really handy for the more routine admin tasks but in my experience it doesn't appear to upgrade very well. Otherwise, it seems quite polished. This is the first time it's taken out my FS config though.

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How to get callid
by Mino Haluz 01 Jul '13

01 Jul '13

Hi, before briding I want to add to custom header callid of inbound call. <action application="set" data="sip_h_X-Origination-Leg=???"/> what should I put there so that it will be filled with this callid? Thanks M

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