July 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

char *exec_on_max_fail | char *exec_on_max_timeout;
by Cesar Bermudez 03 Jul '13

03 Jul '13

Hi List. How can use the exec_on_max_fail and the exec_on_max_timeout in a IVR ? Best regards.

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(no subject)
by mang jose 03 Jul '13

03 Jul '13

Has anyone tried calling from freeswitch to the rtmp client? I can make outgoing calls but can't receive incoming calls, it says the number is unavailable

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Re: [Freeswitch-users] ReINVITE failure with ACK+SDP scenario
by BNML 02 Jul '13

02 Jul '13

Sorry I meant : "I am _now_ able". Thanks you Le 02/07/2013 19:10, BNML a écrit : > I am not able to propagate the ACK+SDP (and the INVITE without SDP) to > the carrier. Thanks ! > > Le mar. 02 juil. 2013 18:13:40 CEST, Shona McNeill a écrit : >> Try adding >> >> <param name="enable-3pcc" value="proxy"/> >> >> to the sip profile. >> >> The horror that is "bypass media with proxy-mode 3pcc" should handle ACK+SDP after a no-SDP invite. :-) >> At the least, it may point the way to what else needs to be added to make it all work. >> >> >> Shona >> >> >> >>> -----Original Message----- >>> From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch- >>> users-bounces(a)lists.freeswitch.org] On Behalf Of BNML >>> Sent: 02 July 2013 16:49 >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] ReINVITE failure with ACK+SDP scenario >>> >>> I am suspecting a patch on mod_sofia migth be needed in order to >>> resolve the Late SDP Negotiation in an RE-INVITE scenario to avoir the >>> one-way audio. I can get involved in writing the patch, if needed. >>> >>> Here is the FS console logs along with sip trace >>> http://pastebin.freeswitch.org/21141 >>> >>> Le 01/07/2013 14:40, Muhammad Shahzad a écrit : >>> >>> >>> Can you send FS console logs along with sip trace enabled? Just >>> curious how FS responds to this situation. >>> >>> Thank you. >>> >>> >>> On Mon, Jul 1, 2013 at 7:21 PM, BNML <bnml(a)andrexen.com> wrote: >>> >>> >>> Hi, >>> >>> The problem is that FreeSWITCH doesn't forward Late SDP >>> negotiation in a RE-INVITE scenario which end up to one way audio. >>> >>> After testing the option "enable-soa" didn't resolved the >>> issue. >>> >>> I want to bridge two calls that are already established (I >>> am not dealing with INVITE but with Re-INVITE) . >>> That's why inbound-late-negotiation (which is within the >>> dialplan) is not in the scope of this (I have it activated already). >>> >>> >>> > When Call B is answered, you should use SDP in 200 OK as >>> basis for SDP of ACK in Call A. Otherwise you will get one way audio as >>> expect. >>> >>> >>> Yes. This is the scenario I described at state 4) of the >>> schema. >>> >>> >>> Thanks >>> >>> BNML >>> >>> Le 06/28/2013 07:00 PM, Muhammad Shahzad a écrit : >>> >>> >>> I am not sure what you are trying to do here. You >>> said we want to originate two calls from your PABX to carrier through >>> FS, i.e. >>> >>> Call A: PABX -> FS -> Carrier -> User A >>> Call B: PABX -> FS -> Carrier -> User B >>> >>> And then bridge them on FS, such that, >>> >>> User A <- Carrier <- FS -> Carrier -> User B >>> >>> I think there are easier ways to achieve this. But >>> anyway in this scenario, >>> >>> Since Call A does not have SDP, which means you are >>> triggering Late SDP Negotiation. Make sure you have enabled in sofia >>> profile, >>> >>> <param name="inbound-late-negotiation" value="true"/> >>> >>> So, that FS does not Answer it with local SDP. >>> Additionally you may need to disable SOA, >>> >>> <param name="enable-soa" value="false"/> >>> >>> When 200 OK for Call A arrives (from carrier, through >>> FS to PABX), you use it as basis for SDP of INVITE in Call B. That's >>> interesting.. >>> >>> When Call B is answered, you should use SDP in 200 OK >>> as basis for SDP of ACK in Call A. Otherwise you will get one way audio >>> as expect. >>> >>> >>> Thank you. >>> >>> >>> >>> >>> On Fri, Jun 28, 2013 at 6:56 PM, BNML >>> <bnml(a)andrexen.com> wrote: >>> >>> >>> 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 >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel >>> Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> -- >>> >>> Mit freundlichen Grüßen >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication >>> Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch- >>> users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> >>> -- >>> >>> Mit freundlichen Grüßen >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +49 176 99 83 10 85 >>> MSN: shari_786pk(a)hotmail.com >>> Email: shaheryarkh(a)googlemail.com >>> >>> >>> >>> _________________________________________________________________ >>> ________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitc >>> h-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >

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adding G729 licences
by Mino Haluz 02 Jul '13

02 Jul '13

Hi, how can I add additional g729 licences in FS. I already have some installed, but after copying new licence file to /etc/freeswitch, it does not make sum, it takes only the last licence file. M

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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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