- FreeSWITCH-users - FreeSWITCH Mailing Lists

ClueCon Weekly! Today at 1PM Eastern!
by Ken Rice 02 Jul '14

02 Jul '14

Join us for ClueCon Weekly @ sip:888@conference.freeswitch.org Todays Topic: Stash! What it means for you and how to use it! Join us today at 1PM Eastern, 10AM Pacific for a preview of whats to come for the FreeSWITCH SCM! And don¹t forget if you havent registered for ClueCon, visit http://cluecon.com and register today! Space is filling up for the Training! So get your space reserved today! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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REMINDER!!!! STASH IS COMING !!!
by Ken Rice 02 Jul '14

02 Jul '14

Hey FreeSWITCHers Stash is coming! The URLs will be updating about 9PM CST Tomorrow July 2nd. Watch for the announcement!! One the cut happens git.freeswitch.org will go read only. It will hang around long enough for everyone to migrate off of it, but you¹ll want to switch to the new Stash! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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mod_fifo out and uuid_simplify?
by George Niculae 02 Jul '14

02 Jul '14

Hi All, I am using mod_fifo for park/unpark like application using diaplan below, everything works great, I am able to park/unpark calls, transfer back to the extension that parked the call on timeout and remove FS from signaling path after transfer using uuid_simplify. What I wasn't able to accomplish is to remove FS from signaling path after unparking the call, tried both uuid_simplify and setting sip_auto_simplify variable without luck. Any idea if this is possible? Thanks, George <extension name="ParkOrbitFull"> <condition field="destination_number" expression="^5555$"> <action application="set" data="fifo_music=${hold_music"/> <action application="set" data="fifo_orbit_exten=transferBack${sip_referred_by_user}:30"/> <action application="set" data="fifo_orbit_dialplan=XML"/> <action application="set" data="fifo_caller_exit_key=5"/> <action application="set" data="fifo_caller_exit_to_orbit=true"/> <action application="fifo" data="QPark5555 in"/> <action application="hangup"/> </condition> </extension> <extension name="UnParkFull"> <condition field="destination_number" expression="^\*45555$"> <action application="fifo" data="QPark5555 out nowait"/> </condition> </extension> <extension name="transferBack"> <condition field="destination_number" expression="^transferBack(\d+)$"> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="{api_on_answer='uuid_simplify ${uuid}'}sofia/$${domain}/$1@$${domain}"/> </condition> </extension>

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Freeswitch Week in Review(Master Branch) June 22nd-28th
by Kathleen King 02 Jul '14

02 Jul '14

Hello, again. This week in the FreeSWITCH master branch we had 23 commits. It was relatively quiet week, but some of the features include a few changes and improvements to mod_sofia and the addition of a freeswitch-timezones debian package. The following bugs were squashed: 1a58f05 force UPnP to bind to "local_ip_v4" (LAN) Jira: http://jira.freeswitch.org/browse/FS-6621 ad1e7e9 Fixed updating of the modem type after a FAX ECM CTC in spandsp b9a51bb FS-6632 fixed bug where trying to actually use opus in stereo broke pretending to use it in stereo Jira: http://jira.freeswitch.org/browse/FS-6632 25c0f7d FS-6440 fixed regression causing startup race on single CPU or VM machines Jira: http://jira.freeswitch.org/browse/FS-6440 c71f89a fixed a bug causing a core dump after startup in mod_shout Jira: http://jira.freeswitch.org/browse/FS-6627 557f1d0 Fixed issue handling modem renegotiation when a T.30 CTC message is received in spandsp New features that were added: 80ed14a Ensure mod_sofia params can be unset or reset in mod_sofia d5fa284 Allow setting format of log filename in format_cdr 52fb53c Allow reincarnation from mod_sofia's shutdown-on-fail in mod_sofia f31641f Allow more SRTP errors before killing call af4f104 Create freeswitch-timezones debian package 7741f40 Handle SRTP_READ_ERROR in mod_rayo f7b4ec6 Allow SIP UPDATE messages without display updates in mod_sofia 7406be6 Relay cause of hangup on SRTP failure Improvements in cross platform build supports: 6540068 Fixed a compiler warning in mod_erlang_event In terms of stability these were the use cases that were fixed: 75feed6 Avoid leaking memory on sofia profile reload in mod_sofia Feedback is welcome at kathleen.king(a)quentustech.com and the commits for this week are in the attached file along with the corresponding Jira links. -- Kathleen King Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Cell: (703) 859-3757 kathleen.king(a)quentustech.com

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Stash is Coming! Stash is Coming!
by Ken Rice 01 Jul '14

01 Jul '14

Hey FreeSWITCHers, This is a heads up announcement! The FreeSWITCH Project would like to announce a big change to our main Source Control Repositories. We are about to finish a migration from gitolit for managing the git repos over to Atlassian¹s Stash. What does this mean for the average FreeSWITCH user? Not much really. The biggest impact will be those that use the git repos will need to update their repo remote URLs. Currently this change is slated to happen the evening of Wed July 2 US Central Time. We will be publishing a doc very shortly on what changes you will have to make to your current git checkouts. Any questions let us know! Thanks K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch
by Stanislav Sinyagin 01 Jul '14

01 Jul '14

Kees, do I understand it correctly that you're only sending audio towards the SIP endpoint, so it's not a bidirectional conversation? In this case, you can utilize audio streaming via HTTP, probably it's the easiest way to deliver audio toward FreeSWITCH: session.streamFile("shout://some.server.com/file.mp3", "") https://wiki.freeswitch.org/wiki/Mod_shout ________________________________ From: Kees Jan Koster <kjkoster(a)java-monitor.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>; ssinyagin(a)yahoo.com Sent: Tuesday, July 1, 2014 1:27 PM Subject: Re: [Freeswitch-users] handling signalling externally, RTP via freeswitch Dear Stanislav, I was rather hoping to avoid that path. My C is not all that great. :-/ But if I have to... Any endpoints that come close that I might want to look at? Kees Jan > AFAIK the RTP parameters and streaming are handled from the endpoint module: > https://wiki.freeswitch.org/wiki/Modules > > so, you might end up in writing your own module for what you want. > > > From: Kees Jan Koster <kjkoster(a)kjkoster.org> > To: freeswitch-users(a)lists.freeswitch.org > Sent: Tuesday, July 1, 2014 9:21 AM > Subject: [Freeswitch-users] handling signalling externally, RTP via freeswitch > > Dear All, > > I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge. > > I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side. > > Call flow (A-leg is the custom client, B-leg is sofia-SIP) > [1] custom client starts call to SIP side over ESL [1] > FreeSwitch handles the signalling with the SIP client > SIP client answers the call > [2] FreeSwitch sends event which tells the client where the RTP stream should go to > custom client starts sending RTP audio > FreeSwitch bridges the two legs and people talk > either side hangs up and the call is done > > The reverse (SIP dialling the custom client) won't happen and does not have to be supported. > > My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below? > > bgapi originate [1] sofia/internal/1002(a)example.com > > -- > Kees Jan > > http://java-monitor.com/ > kjkoster(a)kjkoster.org > +31651838192 > > The secret of success lies in the stability of the goal. -- Benjamin Disraeli > > > _________________________________________________________________________ > 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 > > > _________________________________________________________________________ > 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 -- Kees Jan kjkoster(a)java-monitor.com http://java-monitor.com/ +31651838192 The secret of success lies in the stability of the goal. -- Benjamin Disraeli

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Error while installing freeswitch
by Saghar Ayyaz 01 Jul '14

01 Jul '14

Hello, I am getting following error while installing freeswitch on Ubuntu 14.04 LTS. libtool: link: cannot find the library /usr/src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la' or unhandled argument/usr/src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la' I read from a forum that this problem can be solved if I install lafilefixer and run "lafilefixer --justfixit". But I don't find a any way to install lafilefixer. Can somebody help me to solve this issue? Best Regards, Saghar Ayyaz

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handling signalling externally, RTP via freeswitch
by Kees Jan Koster 01 Jul '14

01 Jul '14

Dear All, I need some reading advice/pointers on the following: I want to use FreeSwitch as a bridge from a custom client to SIP. The SIP part is easy and works. The non-SIP part is the challenge. I have my audio sending client hooked up via mod_event_socket. I really like the way events and ESL work in FreeSwitch, by the way. Good control and easy to code. But I digress. So I have my client hooked up over ESL for signalling and need to know where to send the RTP stream to from the non-SIP client side. Call flow (A-leg is the custom client, B-leg is sofia-SIP) [1] custom client starts call to SIP side over ESL [1] FreeSwitch handles the signalling with the SIP client SIP client answers the call [2] FreeSwitch sends event which tells the client where the RTP stream should go to custom client starts sending RTP audio FreeSwitch bridges the two legs and people talk either side hangs up and the call is done The reverse (SIP dialling the custom client) won't happen and does not have to be supported. My concrete question is how [1] should happen. The custom client should issue an "bgapi originate". The arguments for the SIP side are clear, but what to use for the custom client? What should I fill in for [1], below? bgapi originate [1] sofia/internal/1002(a)example.com -- Kees Jan http://java-monitor.com/ kjkoster(a)kjkoster.org +31651838192 The secret of success lies in the stability of the goal. -- Benjamin Disraeli

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Re: [Freeswitch-users] Recording in flac format ?
by julien terrasson 01 Jul '14

01 Jul '14

Hi Brian, I'm still interested in recording in flac format but i'm not very familiar with mod_sndfile.c (and not familiar with c programming either). To get started, i tried to introduce the flac extension the same way oga extension was introduced : } else if (!strcmp(ext, "oga")) { context->sfinfo.format = SF_FORMAT_OGG | SF_FORMAT_VORBIS; context->sfinfo.samplerate = handle->samplerate; } else if (!strcmp(ext, "flac")) { context->sfinfo.format = SF_FORMAT_FLAC | SF_FORMAT_PCM_16; context->sfinfo.samplerate = handle->samplerate; } And i also added the extension in the list : char *extras[] = { "r8", "r16", "r24", "r32", "gsm", "ul", "ulaw", "al", "alaw", "adpcm", "vox","flac", NULL }; then make & make install. But my first attempt was not really successfull : 2014-07-01 11:47:36.670463 [ERR] mod_sndfile.c:205 Error Opening File [/var/callid:84/record0.flac] [File contains data in an unimplemented format.] Does anybody have docs/guidelines on how to use libsndfile ? Regards, J.terrasson ..Adding formats would need you to modify mod_sndfile to enable them. --- Brian West From: julien terrasson <julien.terrasson(a)gmail.com> To: freeswitch-users(a)lists.freeswitch.org Cc: Date: Wed, 18 Jun 2014 14:58:22 +0200 Subject: [Freeswitch-users] Fwd: Recording in flac format ? Hi, I'm using freeswitch version 1.4.5 I would like to record a session in .flac format. I assume this could be somehow possible since FS rely on libsndfile-1.0.26pre5 and libsndfile claims flac support since v1.0.18. I tried to compiled FS with instruction from description of https://jira.freeswitch.org/browse/FS-1197 : Apt-get install libflac-dev and "./configure --enable-flac".. However that didn't help, flac extension is still not usable on FS recording app. Does anyone have an idea how to have flac format enabled ? Regards, J.Terrasson

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RTCP and OPUS
by Kamil Nigmatullin 01 Jul '14

01 Jul '14

Hello all. What I basicly want is to have real adaptive codec on both sides (client and FS), that could exchange link quality information through RTCP. As far as I see, after a brief looking through some parts of source code of FS, mod_opus works as static codec and don't change its behavior when Internet link is changing because it has no communication with RTCP module. Could anybody of developers recommend me the most efficient way to deploy this communication process through Freeswitch Core? Is there a way to export and import information from/to module/core during the call? Thanks in advance.

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