From msc at freeswitch.org Wed Dec 1 09:20:13 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Dec 2010 09:20:13 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_01 We'll be doing another community documentation session but first we'll have Chad Phillips, aka hunmonk, talking to us about his Jester Mail project. Talk to you in a bit. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101201/991cf413/attachment.html From msc at freeswitch.org Wed Dec 1 12:16:51 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Dec 2010 12:16:51 -0800 Subject: [Freeswitch-dev] FreeSWITCH "Staff" Conference Call on Thursday Dec 2 Message-ID: Hello all, We are planning a 30 minute staff conf call for tomorrow, December 2 at 1PM EST / 10AM PST. If you are willing and able to assist with the FreeSWITCH infrastructure maintenance and are ready to roll up your sleeves and do some work then please join us tomorrow. We will use the FreeSWITCH public conference: SIP: 888 at conference.freeswitch.org PSTN: 1-919-386-9900 Skype: skypiax5 Some of the things we will be discussing: Who can do what? (i.e. what are you good at and what would you like to learn how to do) FS infrastructure overview: what we have, what we are planning to do, etc. and stuff that needs management: * git * JIRA * Fisheye * Drupal sites (freeswitch.org, cluecon.com, ostag.org) * Mailing lists (-users, -dev, etc.) * other servers (db, conf box, etc.) We have a lot to go over, but initially we just need to get everything out on the table. If you want to help please email me off list and provide me your contact information and what your skills are. We are filling out this list: http://wiki.freeswitch.org/wiki/Community_Staff Thank you all for supporting FreeSWITCH! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101201/dcb8c227/attachment.html From alexander.lyasin at gmail.com Fri Dec 3 08:42:50 2010 From: alexander.lyasin at gmail.com (Alexander Lyasin) Date: Fri, 3 Dec 2010 10:42:50 +0500 Subject: [Freeswitch-dev] FreeSWITCH module controlled by external application Message-ID: <201012031042.50611.alexander.lyasin@gmail.com> Hello all! I interests the next questions: Are Possible to use FreeWSITCH how UA? I have a task to write FreeSWITCH module (for Linux) that get an information (a signals) from another application and establishes SIP transactions (with FreeSWITCH functions) with another SIP agents. Ex: An Application sends a signal to the FreeSWITCH module (eg using a message queue), transmits the data about the caller and the called party. The module should to establish the session between them with FreeSWITCH functions. In this case voice will not be transmitted with FreeSWITCH. Perhaps this problem has been resolved already. Can are you provide me a link to a similar case, examples or documentation, how it done within FreeSWITCH (preferably a C )? Thank you very much! From cmrienzo at gmail.com Fri Dec 3 18:15:47 2010 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Fri, 3 Dec 2010 10:15:47 -0500 Subject: [Freeswitch-dev] FreeSWITCH module controlled by external application In-Reply-To: <201012031042.50611.alexander.lyasin@gmail.com> References: <201012031042.50611.alexander.lyasin@gmail.com> Message-ID: There are many possibilities with FreeSWITCH. One option is to control FreeSWITCH over the event socket. See http://wiki.freeswitch.org/mod_event_socket for the protocol description and http://wiki.freeswitch.org/esl for a client library to connect to the event socket. On Fri, Dec 3, 2010 at 12:42 AM, Alexander Lyasin < alexander.lyasin at gmail.com> wrote: > Hello all! > I interests the next questions: > > Are Possible to use FreeWSITCH how UA? > I have a task to write FreeSWITCH module (for Linux) that get an > information > (a signals) from another application and establishes SIP transactions (with > FreeSWITCH functions) with another SIP agents. > > Ex: > An Application sends a signal to the FreeSWITCH module (eg using a message > queue), transmits > the data about the caller and the called party. The module should to > establish > the session between them with FreeSWITCH functions. In this case voice > will > not be transmitted with FreeSWITCH. > Perhaps this problem has been resolved already. Can are you provide me a > link > to a similar case, examples or documentation, how it done within FreeSWITCH > (preferably a C )? > Thank you very much! > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101203/0593af2c/attachment.html From cmrienzo at gmail.com Fri Dec 3 18:16:43 2010 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Fri, 3 Dec 2010 10:16:43 -0500 Subject: [Freeswitch-dev] FreeSWITCH module controlled by external application In-Reply-To: References: <201012031042.50611.alexander.lyasin@gmail.com> Message-ID: Fixing my broken links... http://wiki.freeswitch.org/wiki/mod_event_socket http://wiki.freeswitch.org/wiki/esl On Fri, Dec 3, 2010 at 10:15 AM, Christopher Rienzo wrote: > There are many possibilities with FreeSWITCH. One option is to control > FreeSWITCH over the event socket. See > http://wiki.freeswitch.org/mod_event_socket for the protocol description > and http://wiki.freeswitch.org/esl for a client library to connect to the > event socket. > > > > On Fri, Dec 3, 2010 at 12:42 AM, Alexander Lyasin < > alexander.lyasin at gmail.com> wrote: > >> Hello all! >> I interests the next questions: >> >> Are Possible to use FreeWSITCH how UA? >> I have a task to write FreeSWITCH module (for Linux) that get an >> information >> (a signals) from another application and establishes SIP transactions >> (with >> FreeSWITCH functions) with another SIP agents. >> >> Ex: >> An Application sends a signal to the FreeSWITCH module (eg using a message >> queue), transmits >> the data about the caller and the called party. The module should to >> establish >> the session between them with FreeSWITCH functions. In this case voice >> will >> not be transmitted with FreeSWITCH. >> Perhaps this problem has been resolved already. Can are you provide me a >> link >> to a similar case, examples or documentation, how it done within >> FreeSWITCH >> (preferably a C )? >> Thank you very much! >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101203/c5fe5aba/attachment.html From raison at chatsubo.net Mon Dec 6 09:40:06 2010 From: raison at chatsubo.net (Kevin Raison) Date: Sun, 05 Dec 2010 22:40:06 -0800 Subject: [Freeswitch-dev] playback on uuid_bridge Message-ID: <4CFC8546.1020902@chatsubo.net> I would like to play a wav file on only one leg of a bridged call (setup using uuid_bridge) in order to keep the message private to that leg. Using "playback" plays the message on both legs. Is there a way to make the playback private? Thanks, Kevin Raison From lakindia89 at gmail.com Mon Dec 6 09:50:43 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Mon, 6 Dec 2010 12:20:43 +0530 Subject: [Freeswitch-dev] playback on uuid_bridge In-Reply-To: <4CFC8546.1020902@chatsubo.net> References: <4CFC8546.1020902@chatsubo.net> Message-ID: Check the uuid_broadcast and uuid_displace on the wiki. On Mon, Dec 6, 2010 at 12:10 PM, Kevin Raison wrote: > I would like to play a wav file on only one leg of a bridged call (setup > using uuid_bridge) in order to keep the message private to that leg. > Using "playback" plays the message on both legs. Is there a way to make > the playback private? > > Thanks, > Kevin Raison > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/052d0513/attachment-0001.html From d.mordovin at dwide.com Mon Dec 6 12:53:48 2010 From: d.mordovin at dwide.com (Dmitry Mordovin) Date: Mon, 06 Dec 2010 12:53:48 +0300 Subject: [Freeswitch-dev] Bridged call duration Message-ID: <4CFCB2AC.1050609@dwide.com> Hello All! Can't find 'bridge' options to set maximum call duration. There is a way to do that? Best regards, Dmitry Mordovin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/0888a2f7/attachment.html From dujinfang at gmail.com Mon Dec 6 13:49:25 2010 From: dujinfang at gmail.com (Seven Du) Date: Mon, 6 Dec 2010 18:49:25 +0800 Subject: [Freeswitch-dev] Bridged call duration In-Reply-To: <4CFCB2AC.1050609@dwide.com> References: <4CFCB2AC.1050609@dwide.com> Message-ID: sched_api or sched_hangup? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup On Mon, Dec 6, 2010 at 5:53 PM, Dmitry Mordovin wrote: > Hello All! > > Can't find 'bridge' options to set maximum call duration. > > There is a way to do that? > > Best regards, > Dmitry Mordovin > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From d.mordovin at dwide.com Mon Dec 6 13:56:16 2010 From: d.mordovin at dwide.com (Dmitry Mordovin) Date: Mon, 06 Dec 2010 13:56:16 +0300 Subject: [Freeswitch-dev] Bridged call duration In-Reply-To: References: <4CFCB2AC.1050609@dwide.com> Message-ID: <4CFCC150.4040406@dwide.com> On 12/06/2010 01:49 PM, Seven Du wrote: > sched_api or sched_hangup? > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup > > On Mon, Dec 6, 2010 at 5:53 PM, Dmitry Mordovin wrote: > >> Hello All! >> >> Can't find 'bridge' options to set maximum call duration. >> >> There is a way to do that? >> >> Best regards, >> Dmitry Mordovin >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> >> > > > Hi! Its help me! Thank you! Best regards, Dmitry Mordovin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/76813303/attachment.html From grw.freeswitch at gmail.com Mon Dec 6 19:32:53 2010 From: grw.freeswitch at gmail.com (Geovani Ricardo Wiedenhoft) Date: Mon, 6 Dec 2010 14:32:53 -0200 Subject: [Freeswitch-dev] mod_khomp Message-ID: The Endpoint and the documentation for the Khomp boards (mod_khomp) are available on the official FreeSWTICH branch. Also, the documents are available on the wiki. http://wiki.freeswitch.org/wiki/Khomp The version is compatible with all Khomp boards (SPX series): - FXS - FXO - E1 - R2, R2 DTMF and OpenCAS in Hardware, ISDN (User, Network), OpenCCS, LineSide, CAS_EL7, E1LC - GSM with send/receive SMS (boards and usb devices) - Passive record (R2, ISDN, OpenCAS, OpenCCS and FXO) - kommuter Visit our site to more information about Khomp products: http://www.khomp.com.br Thank you. :) Khomp development team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/80e4573d/attachment.html From raison at chatsubo.net Mon Dec 6 23:58:10 2010 From: raison at chatsubo.net (Kevin Raison) Date: Mon, 06 Dec 2010 12:58:10 -0800 Subject: [Freeswitch-dev] playback on uuid_bridge In-Reply-To: References: <4CFC8546.1020902@chatsubo.net> Message-ID: <4CFD4E62.8000509@chatsubo.net> Thanks you, lakshmanan. That is exactly what I was looking for. One more question related to this one: in that same conference I would like one leg to be able to issue DTMF-based commands without the other leg hearing the DTMF. Is there a call variable that will confine DTMF so that it cannot be heard on the other leg? I see some discussion of this on the wiki DTMF page, but no commands or variables are mentioned. Thanks again! -Kevin On 12/5/10 10:50 PM, lakshmanan ganapathy wrote: > Check the uuid_broadcast and uuid_displace on the wiki. > > > On Mon, Dec 6, 2010 at 12:10 PM, Kevin Raison > wrote: > > I would like to play a wav file on only one leg of a bridged call (setup > using uuid_bridge) in order to keep the message private to that leg. > Using "playback" plays the message on both legs. Is there a way to make > the playback private? > > Thanks, > Kevin Raison > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Kevin Raison Founder, CTO Chatsubo.net, LLC 9708 1st Ave NW Seattle, WA 98117 raison at chatsubo.net ph: +1 (206) 801-5728 fx: +1 (206) 801-5729 From msc at freeswitch.org Tue Dec 7 01:17:20 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Dec 2010 14:17:20 -0800 Subject: [Freeswitch-dev] playback on uuid_bridge In-Reply-To: <4CFD4E62.8000509@chatsubo.net> References: <4CFC8546.1020902@chatsubo.net> <4CFD4E62.8000509@chatsubo.net> Message-ID: Kevin, Just a note: you said "conference" when I think you meant bridge. A "conference" is a completely separate entity in FreeSWITCH. Just confirming: you are talking about a simple bridge between two call legs, correct? -MC On Mon, Dec 6, 2010 at 12:58 PM, Kevin Raison wrote: > Thanks you, lakshmanan. That is exactly what I was looking for. > > One more question related to this one: in that same conference I would > like one leg to be able to issue DTMF-based commands without the other > leg hearing the DTMF. Is there a call variable that will confine DTMF > so that it cannot be heard on the other leg? I see some discussion of > this on the wiki DTMF page, but no commands or variables are mentioned. > > Thanks again! > > -Kevin > > > > On 12/5/10 10:50 PM, lakshmanan ganapathy wrote: > > Check the uuid_broadcast and uuid_displace on the wiki. > > > > > > On Mon, Dec 6, 2010 at 12:10 PM, Kevin Raison > > wrote: > > > > I would like to play a wav file on only one leg of a bridged call > (setup > > using uuid_bridge) in order to keep the message private to that leg. > > Using "playback" plays the message on both legs. Is there a way to > make > > the playback private? > > > > Thanks, > > Kevin Raison > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > -- > Kevin Raison > Founder, CTO > > Chatsubo.net, LLC > 9708 1st Ave NW > Seattle, WA 98117 > raison at chatsubo.net > > ph: +1 (206) 801-5728 > fx: +1 (206) 801-5729 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/95a74a45/attachment.html From raison at chatsubo.net Tue Dec 7 01:25:04 2010 From: raison at chatsubo.net (Kevin Raison) Date: Mon, 06 Dec 2010 14:25:04 -0800 Subject: [Freeswitch-dev] playback on uuid_bridge In-Reply-To: References: <4CFC8546.1020902@chatsubo.net> <4CFD4E62.8000509@chatsubo.net> Message-ID: <4CFD62C0.4080607@chatsubo.net> Yes, sorry, a bridge setup using uuid_bridge is what I mean. On 12/6/10 2:17 PM, Michael Collins wrote: > Kevin, > > Just a note: you said "conference" when I think you meant bridge. A > "conference" is a completely separate entity in FreeSWITCH. Just > confirming: you are talking about a simple bridge between two call legs, > correct? > > -MC > > On Mon, Dec 6, 2010 at 12:58 PM, Kevin Raison > wrote: > > Thanks you, lakshmanan. That is exactly what I was looking for. > > One more question related to this one: in that same conference I would > like one leg to be able to issue DTMF-based commands without the other > leg hearing the DTMF. Is there a call variable that will confine DTMF > so that it cannot be heard on the other leg? I see some discussion of > this on the wiki DTMF page, but no commands or variables are mentioned. > > Thanks again! > > -Kevin > > > > On 12/5/10 10:50 PM, lakshmanan ganapathy wrote: > > Check the uuid_broadcast and uuid_displace on the wiki. > > > > > > On Mon, Dec 6, 2010 at 12:10 PM, Kevin Raison > > > >> wrote: > > > > I would like to play a wav file on only one leg of a bridged > call (setup > > using uuid_bridge) in order to keep the message private to > that leg. > > Using "playback" plays the message on both legs. Is there a > way to make > > the playback private? > > > > Thanks, > > Kevin Raison > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > -- > Kevin Raison > Founder, CTO > > Chatsubo.net, LLC > 9708 1st Ave NW > Seattle, WA 98117 > raison at chatsubo.net > > ph: +1 (206) 801-5728 > fx: +1 (206) 801-5729 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -- Kevin Raison Founder, CTO Chatsubo.net, LLC 9708 1st Ave NW Seattle, WA 98117 raison at chatsubo.net ph: +1 (206) 801-5728 fx: +1 (206) 801-5729 From msc at freeswitch.org Tue Dec 7 01:31:14 2010 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Dec 2010 14:31:14 -0800 Subject: [Freeswitch-dev] playback on uuid_bridge In-Reply-To: <4CFD62C0.4080607@chatsubo.net> References: <4CFC8546.1020902@chatsubo.net> <4CFD4E62.8000509@chatsubo.net> <4CFD62C0.4080607@chatsubo.net> Message-ID: On Mon, Dec 6, 2010 at 2:25 PM, Kevin Raison wrote: > Yes, sorry, a bridge setup using uuid_bridge is what I mean. > > Check out bind_digit_action or bind_meta_app. I've used those in the past to let one leg have dtmf controls. It's been my experience that the other side doesn't hear the DTMFs but I believe I've been doing everything out of band. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101206/0172dc6f/attachment.html From michael at ostag.org Wed Dec 8 04:11:22 2010 From: michael at ostag.org (Michael Collins) Date: Tue, 7 Dec 2010 17:11:22 -0800 Subject: [Freeswitch-dev] OSTAG - How You Can Help Message-ID: Greetings! The OSTAG team would like to let everyone know that we are ready for non-profit work. While monetary donations are always welcomed they certainly are not the only way that you can help. We have added a new section to the ostag.org Web site on how to donate . Please review it and think about how you can donate time, money, or resources to the OSTAG project. All donations are tax-deductible. Thank you for continuing to support Open Source software! -The OSTAG Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101207/3b0c7c5f/attachment.html From joegen at opensipstack.org Wed Dec 8 04:36:41 2010 From: joegen at opensipstack.org (Joegen E. Baclor) Date: Wed, 08 Dec 2010 09:36:41 +0800 Subject: [Freeswitch-dev] Scriptable test-suite Message-ID: <4CFEE129.8030505@opensipstack.org> Hi, We are trying to work out a test suite for sipXecs media services and we thought that freeswitch itself is the best place to do it. I am initiallly thinking of controlling freeswitch using ESL inbound to simulate traversing of sipXecs media services ivr menus. However, although ESL has the most flexibility among the choices, I find the scripting modules really appealing specially because it exposes the ability to add new test scenarios. Am I correct that ESL inbound function(eg: placing independent outbound calls without having to hit a dialplan) is not currently doable via the mod_scripting_language plugins? Because of this, I am plannning to simply expose ESL inbound/outbound library via a google::v8 glue to make the test scenarios somehow scriptable. I am wondering if I am not crazy if i do so. I have came accross this thread old http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/10926/focus=10928 . I am wondeing if more work/thoughts came into this so i wont be reinventing the wheel. advice appreciated. Thanks! Joegen From msc at freeswitch.org Wed Dec 8 18:57:28 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Dec 2010 07:57:28 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Don't forget about today's call! http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_08 Our agenda is light today so bring your questions and suggestions and we'll discuss them as a group. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101208/ba0e93ca/attachment.html From msc at freeswitch.org Thu Dec 9 21:34:30 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Dec 2010 10:34:30 -0800 Subject: [Freeswitch-dev] Scriptable test-suite In-Reply-To: <4CFEE129.8030505@opensipstack.org> References: <4CFEE129.8030505@opensipstack.org> Message-ID: Joegen, I don't know if you already came to a conclusion here but I just wanted to chime in a bit. First off, understand that ESL inbound is extremely powerful - you can do pretty much anything you want. In fact, there is nothing preventing you from connecting in via ESL, then launching a call that goes through the dialplan and then executes a script written in Lua/Perl/Javascript. My advice to you is to break this down into smaller pieces. Create a dialplan extension and script that tests an IVR and make sure it works. Then create an ESL application that controls FreeSWITCH and can launch one or more calls through your test extension. Repeat as needed. -MC On Tue, Dec 7, 2010 at 5:36 PM, Joegen E. Baclor wrote: > Hi, > > We are trying to work out a test suite for sipXecs media services and we > thought that freeswitch itself is the best place to do it. I am > initiallly thinking of controlling freeswitch using ESL inbound to > simulate traversing of sipXecs media services ivr menus. However, > although ESL has the most flexibility among the choices, I find the > scripting modules really appealing specially because it exposes the > ability to add new test scenarios. Am I correct that ESL inbound > function(eg: placing independent outbound calls without having to hit a > dialplan) is not currently doable via the mod_scripting_language > plugins? Because of this, I am plannning to simply expose ESL > inbound/outbound library via a google::v8 glue to make the test > scenarios somehow scriptable. I am wondering if I am not crazy if i do > so. > > I have came accross this thread old > > http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/10926/focus=10928 > . I am wondeing if more work/thoughts came into this so i wont be > reinventing the wheel. > > advice appreciated. Thanks! > > Joegen > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101209/d5792ddc/attachment.html From anthony.minessale at gmail.com Fri Dec 10 00:30:11 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Dec 2010 15:30:11 -0600 Subject: [Freeswitch-dev] User Interface Programmer Job Opportunity Message-ID: If anyone who lives in the US especially [MI WI OK CA] or would like to move to one of those places and is good at UI design. Contact jobs at freeswitch.org with a resume. We need some programmers to work with us on the CudaTel UI. Its a demanding yet rewarding job in software development and you get to help with FreeSWITCH as well. Jquery/Js/Comet type skills are the most important. -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From vipkilla at gmail.com Fri Dec 10 01:07:49 2010 From: vipkilla at gmail.com (vip killa) Date: Thu, 9 Dec 2010 17:07:49 -0500 Subject: [Freeswitch-dev] User Interface Programmer Job Opportunity In-Reply-To: References: Message-ID: What is the UI used for? If you want super advanced UI you should consider incorporating red5 server for realtime UI. I've successfully done this with FS conferences. On Thu, Dec 9, 2010 at 4:30 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > If anyone who lives in the US especially [MI WI OK CA] or would like > to move to one of those places and is good at UI design. > Contact jobs at freeswitch.org with a resume. > We need some programmers to work with us on the CudaTel UI. > > Its a demanding yet rewarding job in software development and you get > to help with FreeSWITCH as well. > > Jquery/Js/Comet type skills are the most important. > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101209/2797fcb0/attachment.html From edpimentl at gmail.com Fri Dec 10 01:22:25 2010 From: edpimentl at gmail.com (EdPimentl) Date: Thu, 9 Dec 2010 17:22:25 -0500 Subject: [Freeswitch-dev] User Interface Programmer Job Opportunity In-Reply-To: References: Message-ID: My thinking... a JQuery UI and leveraging of HTML5 / CSS3 / JS is all you need. I would be glad to do all the layout/mockup and psd to html,, with valid seo coding and tested for all browsers. Sincerely, -E Gpro.ws On Thu, Dec 9, 2010 at 5:07 PM, vip killa wrote: > What is the UI used for? If you want super advanced UI you should consider > incorporating red5 server for realtime UI. I've successfully done this with > FS conferences. > > > On Thu, Dec 9, 2010 at 4:30 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> If anyone who lives in the US especially [MI WI OK CA] or would like >> to move to one of those places and is good at UI design. >> Contact jobs at freeswitch.org with a resume. >> We need some programmers to work with us on the CudaTel UI. >> >> Its a demanding yet rewarding job in software development and you get >> to help with FreeSWITCH as well. >> >> Jquery/Js/Comet type skills are the most important. >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101209/3f1abda3/attachment-0001.html From anthony.minessale at gmail.com Fri Dec 10 02:14:06 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Dec 2010 17:14:06 -0600 Subject: [Freeswitch-dev] User Interface Programmer Job Opportunity In-Reply-To: References: Message-ID: As in, we are looking for someone to hire permenantly for the CudaTel team On Dec 9, 2010 4:23 PM, "EdPimentl" wrote: > My thinking... a JQuery UI and leveraging of HTML5 / CSS3 / JS is all you > need. > > I would be glad to do all the layout/mockup and psd to html,, with valid seo > coding and tested for all browsers. > > > Sincerely, > -E > Gpro.ws > > > > > On Thu, Dec 9, 2010 at 5:07 PM, vip killa wrote: > >> What is the UI used for? If you want super advanced UI you should consider >> incorporating red5 server for realtime UI. I've successfully done this with >> FS conferences. >> >> >> On Thu, Dec 9, 2010 at 4:30 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> If anyone who lives in the US especially [MI WI OK CA] or would like >>> to move to one of those places and is good at UI design. >>> Contact jobs at freeswitch.org with a resume. >>> We need some programmers to work with us on the CudaTel UI. >>> >>> Its a demanding yet rewarding job in software development and you get >>> to help with FreeSWITCH as well. >>> >>> Jquery/Js/Comet type skills are the most important. >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com < MSN%3Aanthony_minessale at hotmail.com > >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org < sip%3A888 at conference.freeswitch.org > >>> googletalk:conf+888 at conference.freeswitch.org > >>> pstn:+19193869900 >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101209/329863f4/attachment.html From msc at freeswitch.org Fri Dec 10 10:43:04 2010 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Dec 2010 23:43:04 -0800 Subject: [Freeswitch-dev] Asterisk migrations Message-ID: Hey gang, I was just curious if anyone out there had created any tools to assist with migrating from Asterisk (or Trixbox, etc.) to FreeSWITCH. We're starting to see this scenario more and I would like to find out how our community members are handling it. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101209/d0250d8d/attachment.html From lakindia89 at gmail.com Fri Dec 10 10:56:13 2010 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Fri, 10 Dec 2010 13:26:13 +0530 Subject: [Freeswitch-dev] ftmod_libpri - Channel Selection patch In-Reply-To: References: Message-ID: Hi all, Can somebody confirm that the patch is correct?? On Wed, Nov 17, 2010 at 4:34 PM, lakshmanan ganapathy wrote: > I've posted a bug and given the patch as resolution. > > http://jira.freeswitch.org/browse/OPENZAP-118 > > > On Wed, Nov 17, 2010 at 10:43 AM, Moises Silva wrote: > >> Please put this in Jira (jira.freeswitch.org) >> >> Moises Silva >> Senior Software Engineer >> Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R >> 9R6 Canada >> t. 1 905 474 1990 x128 | e. moy at sangoma.com >> >> >> On Tue, Nov 16, 2010 at 10:49 PM, lakshmanan ganapathy < >> lakindia89 at gmail.com> wrote: >> >>> Hi all, >>> When using ftmod_libpri, if a incoming call comes without "Channel >>> Identification" ( in SETUP packet ), then the call didn't get proceeded. >>> I've taken some code snippets from ftmod_isdn and I applied it in >>> ftmod_libpri. I've tested it and in my setup it works fine. >>> >>> I've attached the patch here. Please verify it and found ok commit it. >>> >>> regards, >>> Lakshmanan G. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-dev mailing list >>> FreeSWITCH-dev at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >>> http://www.freeswitch.org >>> >>> >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101210/a2b35a98/attachment.html From joegen at opensipstack.org Fri Dec 10 13:29:45 2010 From: joegen at opensipstack.org (Joegen E. Baclor) Date: Fri, 10 Dec 2010 18:29:45 +0800 Subject: [Freeswitch-dev] Scriptable test-suite In-Reply-To: References: <4CFEE129.8030505@opensipstack.org> Message-ID: <4D020119.4060606@opensipstack.org> Hi Mike, Thanks for the advise. It's good to know i'm on the right track. On Friday, 10 December, 2010 02:34 AM, Michael Collins wrote: > Joegen, > > I don't know if you already came to a conclusion here but I just > wanted to chime in a bit. First off, understand that ESL inbound is > extremely powerful - you can do pretty much anything you want. In > fact, there is nothing preventing you from connecting in via ESL, then > launching a call that goes through the dialplan and then executes a > script written in Lua/Perl/Javascript. > > My advice to you is to break this down into smaller pieces. Create a > dialplan extension and script that tests an IVR and make sure it > works. Then create an ESL application that controls FreeSWITCH and can > launch one or more calls through your test extension. Repeat as needed. > > -MC > > On Tue, Dec 7, 2010 at 5:36 PM, Joegen E. Baclor > > wrote: > > Hi, > > We are trying to work out a test suite for sipXecs media services > and we > thought that freeswitch itself is the best place to do it. I am > initiallly thinking of controlling freeswitch using ESL inbound to > simulate traversing of sipXecs media services ivr menus. However, > although ESL has the most flexibility among the choices, I find the > scripting modules really appealing specially because it exposes the > ability to add new test scenarios. Am I correct that ESL inbound > function(eg: placing independent outbound calls without having to > hit a > dialplan) is not currently doable via the mod_scripting_language > plugins? Because of this, I am plannning to simply expose ESL > inbound/outbound library via a google::v8 glue to make the test > scenarios somehow scriptable. I am wondering if I am not crazy > if i do so. > > I have came accross this thread old > http://thread.gmane.org/gmane.comp.telephony.freeswitch.user/10926/focus=10928 > . I am wondeing if more work/thoughts came into this so i wont be > reinventing the wheel. > > advice appreciated. Thanks! > > Joegen > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101210/df2cb4af/attachment.html From moises.silva at gmail.com Sat Dec 11 05:56:10 2010 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 10 Dec 2010 21:56:10 -0500 Subject: [Freeswitch-dev] ftmod_libpri - Channel Selection patch In-Reply-To: References: Message-ID: On Fri, Dec 10, 2010 at 2:56 AM, lakshmanan ganapathy wrote: > Hi all, > Can somebody confirm that the patch is correct?? > > you should joint irc in channel #freetdm You can talk to me or even better to stkn, he has been the one doing more work in libpri module. Moises Silva Senior Software Engineer Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6 Canada t. 1 905 474 1990 x128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101210/bc095eb4/attachment-0001.html From betergreen at live.com Fri Dec 10 10:40:00 2010 From: betergreen at live.com (peter_green lion) Date: Fri, 10 Dec 2010 14:40:00 +0700 Subject: [Freeswitch-dev] mod_python error Message-ID: hi all, I see mod_python in wiki and i try to make some example as page: http://wiki.freeswitch.org/wiki/Examples_directory_py i have configure in dial plan as: and in /usr/local/freeswitch/scripts/ add file directory.py as: import sys, time, sqlite3 from freeswitch import * digitpath = "/usr/local/freeswitch/sounds/en/us/callie/digits/8000/" custom_sounds_path = "/usr/local/freeswitch/sounds/custom/" def checkforgreeting(extension): conn = sqlite3.connect("/usr/local/freeswitch/db/voicemail_default.db") c = conn.cursor() c.execute("select name_path from voicemail_prefs where username=?", (extension,) ) # retrieve recorded_name path row=c.fetchone() console_log("alert", "row: %s\n" % (str(row))) c.close() if row[0]: return row else: return False def handler(uuid): alphabet = "abcdefghijklmnopqrstuvwxyz" numbers = "22233344455566677778889999" code_to_name = {} code = "" names = { "Allen, Larry": 1000, "Monroe, Beckey": 1001, } lnames = {} for name in names: lnames[name.lower()] = names[name] names = lnames def sayname(fullname): console_log("alert", "Now saying: " + fullname + "\n") # flip first and last... remove comma split_fullname = fullname.split(',') fname_lname = split_fullname[1].lstrip() + " " + split_fullname[0] # take the chars of a name and say each one session.execute("phrase", "spell," + fname_lname); # preprocessing before evaluating arg input # build codes from names dict for name in names: name3char = name[0:3] # empty out the code var code = "" for char in name3char: code = code + numbers[alphabet.index(char)] # code is the 3 digits code generated from the first 3 chars of the last name if not code in code_to_name: code_to_name[code] = [ name ] else: code_to_name[code].append(name) session = PySession(uuid) session.answer() session.execute( "sleep", "2000" ) digits_keyed = session.playAndGetDigits(3, 3, 10, 5000, "*#", custom_sounds_path + "dir-intro.wav", "", ""); # evaluate input # we want 3 digits console_log("alert", "digits_keyed: %s\n" % ( str(digits_keyed) )) if len(digits_keyed) == 3: # it must be in the code db generated from last names earlier if digits_keyed in code_to_name: console_log("alert", "Yes: %s\n" % (str(code_to_name[digits_keyed]))) if len(code_to_name[digits_keyed]) == 1: # only one extension matches console_log("alert", "Extension found: %s\n" % ( str(names[code_to_name[digits_keyed][0]]))) # transfer to the extension extension = str(names[code_to_name[digits_keyed][0]]) recorded_name = checkforgreeting(extension) if recorded_name: console_log("alert", "Saying recorded name\n") session.streamFile( str(recorded_name[0]) ) else: sayname(code_to_name[digits_keyed][0]) session.execute("phrase", "spell," + extension); session.execute( "sleep", "1000" ) # give option of if correct to press 1 otherwise * and start over digits_keyed = session.playAndGetDigits(1, 1, 3, 2000, "#", custom_sounds_path + "dir-instr.wav", "", "1|\*"); console_log("alert", "digits_keys: %s\n" % ( digits_keyed )) if digits_keyed == "1": session.transfer( extension, "XML", "default") else: if digits_keyed == "*": session.streamFile( custom_sounds_path + "dir-nomatch.wav" ) handler(uuid) # session.transfer( "777", "XML", "default") # session.hangup("1") else: # we matched more than one name for item in code_to_name[digits_keyed]: console_log("alert", "Found more than one extension: %s\n" % ( str(names[item]))) # say each one and give option of if not that one to continue console_log("alert", "item: %s\n" % (item)) extension = str(names[item]) recorded_name = checkforgreeting(extension) if recorded_name: session.streamFile( str(recorded_name[0]) ) else: sayname(item) session.execute("phrase", "spell," + extension); session.execute( "sleep", "1000" ) # give option of if correct to press 1 otherwise * and start over digits_keyed = session.playAndGetDigits(1, 1, 3, 2000, "#", custom_sounds_path + "dir-instr.wav", "", "1|\*"); console_log("alert", "digits_keys: %s\n" % ( digits_keyed )) if digits_keyed == "1": session.transfer( extension, "XML", "default") # session.hangup("1") session.streamFile( custom_sounds_path + "dir-nomore.wav" ) handler(uuid) # session.transfer("777", "XML", "default") # session.hangup("1") else: # no valid extension found so transfer back session.streamFile( custom_sounds_path + "dir-nomatch.wav" ) handler(uuid) # session.transfer("777", "XML", "default") # session.hangup("1") session.hangup("1") i log in soft phone and make call to :1234 ==========> call hangup and i see log in freeswitch -c: 2010-12-10 02:35:55.711484 [ERR] mod_python.c:200 Error calling python script TypeError: handler() takes exactly 1 argument (2 given) Traceback (most recent call last): File "", line 1, in NameError: name 'python_makes_sense' is not defined please suggest to fax it and make it work. another thing: if i have mysql database, how can i get value from mysql. example: i have table test in fusionpbx database. mysql> select * from test; +----+------+----------+-------------+ | id | user | password | ms | +----+------+----------+-------------+ | 1 | 123 | 12345678 | 87546345634 | +----+------+----------+-------------+ so: how can i get value password = 12345678 when i enter user =123 (in python code). thanks all so much -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101210/49be197b/attachment.html From msc at freeswitch.org Wed Dec 15 19:55:50 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 15 Dec 2010 08:55:50 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hello all, Here's today's agenda: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_15 Chad Phillips is doing a follow up on Jester Mail and his Lua IVR toolkit. Talk to you all in an hour! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101215/cd8979ed/attachment.html From gabe at gundy.org Wed Dec 15 20:56:52 2010 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 15 Dec 2010 10:56:52 -0700 Subject: [Freeswitch-dev] mod_python error In-Reply-To: References: Message-ID: On Fri, Dec 10, 2010 at 12:40 AM, peter_green lion wrote: > I see mod_python in wiki and i try to make some example as page: > http://wiki.freeswitch.org/wiki/Examples_directory_py That script looks out dated. See the docs here: http://wiki.freeswitch.org/wiki/Mod_python#Python_module_specification > def handler(uuid): > alphabet = "abcdefghijklmnopqrstuvwxyz" > numbers = "22233344455566677778889999" > code_to_name = {} > code = "" Looks the the session should have been passed in there like so: def handler(session, args): > i log in soft phone and make call to :1234 > call hangup and i see log in freeswitch -c: > 2010-12-10 02:35:55.711484 [ERR] mod_python.c:200 Error calling python > script > TypeError: handler() takes exactly 1 argument (2 given) Redefine the handler function and let us know what you find. My advice would be to start with a small script and make sure you understand how it's being called. Something like hello_world. Once you have that working, move on from there. Good luck, Gabe BTW, it seems like this belongs on the regular FreeSWITCH users list. From vicentini.paulo at gmail.com Sat Dec 18 00:20:59 2010 From: vicentini.paulo at gmail.com (Paulo Vicentini) Date: Fri, 17 Dec 2010 19:20:59 -0200 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia Message-ID: Hi, I would like to override the scheme used for digest authorization Actually it is using the scheme coming from sip_www_authenticate_t in the sofia_reg_handle_sip_r_challenge function, ignoring scheme set in xml configuration I would like something like: if(gateway->register_scheme) scheme = gateway->register_scheme; before nua_authenticate(... I am patching sofia to accept HA1 Regards Paulo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101217/f7ce0fe9/attachment.html From steveayre at gmail.com Sat Dec 18 00:29:38 2010 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 17 Dec 2010 21:29:38 +0000 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: References: Message-ID: If you're looking to store passwords encrypted, then that is already supported. Search the Wiki for a1-hash. -Steve On 17 December 2010 21:20, Paulo Vicentini wrote: > Hi, > I would like to override the scheme used for digest authorization > Actually it is using the scheme coming from sip_www_authenticate_t in the > sofia_reg_handle_sip_r_challenge function, ?ignoring scheme set in xml > configuration > > I would like something like: > if(gateway->register_scheme) > ? ??scheme = gateway->register_scheme; > before nua_authenticate(... > I am patching sofia to accept HA1 > Regards > Paulo > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > From vicentini.paulo at gmail.com Sat Dec 18 00:47:34 2010 From: vicentini.paulo at gmail.com (Paulo Vicentini) Date: Fri, 17 Dec 2010 19:47:34 -0200 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: References: Message-ID: Hi Steve Yes, it is all about a1-hash But I did not see support for storing HA1 for a gateway (UAC), even in sofia-lib So that both sofia-lib and freeswitch would need to be patched for that aim Regards Paulo On Fri, Dec 17, 2010 at 7:29 PM, Steven Ayre wrote: > If you're looking to store passwords encrypted, then that is already > supported. > > Search the Wiki for a1-hash. > > -Steve > > > > On 17 December 2010 21:20, Paulo Vicentini > wrote: > > Hi, > > I would like to override the scheme used for digest authorization > > Actually it is using the scheme coming from sip_www_authenticate_t in the > > sofia_reg_handle_sip_r_challenge function, ignoring scheme set in xml > > configuration > > > > I would like something like: > > if(gateway->register_scheme) > > scheme = gateway->register_scheme; > > before nua_authenticate(... > > I am patching sofia to accept HA1 > > Regards > > Paulo > > > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101217/e7b8dfcb/attachment-0001.html From steveayre at gmail.com Sat Dec 18 15:09:38 2010 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 18 Dec 2010 12:09:38 +0000 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: References: Message-ID: Oh ok... yes the a1-hash is in the user directory for people authenticating to FS... not for FS registering outwards to gateways. I remember coming across this in the past and did take a quick look at how to implement it. The main issue I found was that A1 contains the realm, and the realm is provided by the gateway in the 407 response. You don't therefore know the realm needed at the time you generate the A1. Yes, it would be possible to do by finding the realm the gateway is using and generating the A1 from that, but if the gateway changes the realm the A1 will no longer be valid and FS will start failing to authenticate. -Steve On 17 December 2010 21:47, Paulo Vicentini wrote: > Hi Steve > Yes, it is all about a1-hash > But I did not see support for storing HA1 for a gateway (UAC), even in > sofia-lib > So that both sofia-lib and freeswitch would need to be patched for that aim > Regards > Paulo > > On Fri, Dec 17, 2010 at 7:29 PM, Steven Ayre wrote: >> >> If you're looking to store passwords encrypted, then that is already >> supported. >> >> Search the Wiki for a1-hash. >> >> -Steve >> >> >> >> On 17 December 2010 21:20, Paulo Vicentini >> wrote: >> > Hi, >> > I would like to override the scheme used for digest authorization >> > Actually it is using the scheme coming from sip_www_authenticate_t in >> > the >> > sofia_reg_handle_sip_r_challenge function, ?ignoring scheme set in xml >> > configuration >> > >> > I would like something like: >> > if(gateway->register_scheme) >> > ? ??scheme = gateway->register_scheme; >> > before nua_authenticate(... >> > I am patching sofia to accept HA1 >> > Regards >> > Paulo >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-dev mailing list >> > FreeSWITCH-dev at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> > http://www.freeswitch.org >> > >> > >> >> _______________________________________________ >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > From marty at maui-systems.co.uk Tue Dec 14 22:22:51 2010 From: marty at maui-systems.co.uk (Marty Lee) Date: Tue, 14 Dec 2010 19:22:51 +0000 Subject: [Freeswitch-dev] Freeswitch & Solaris 10 Message-ID: <9A8DB267-9125-4C73-9A4B-519BF6AEBDC1@maui-systems.co.uk> Hi, just playing with building FreeSwitch on a Solaris 10 box and am hitting a link error when building the target '.libs/freeswitch' Undefined first referenced symbol in file make_mask32 freeswitch-switch.o ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch Looking at the source, make_mask32 seems to be defined in the spandsp/src/bit_operations.c file, which isn't involved in the linking at this point. Building with Sun Studio 12, just in case it's of any interest. I'll try and move the make_mask32 function in to the library and see if that cures the problem. Thought I'd ask here, as this kind of thing should appear on most platforms if it really is a problem - meaning it might be something at my end. m ----- Marty Lee e: marty at maui-systems.co.uk Technical Director v: +44 845 869 2661 Maui Systems Ltd f: +44 871 433 8922 Scotland, UK w: http://www.maui-systems.co.uk From vicentini.paulo at gmail.com Sat Dec 18 17:35:41 2010 From: vicentini.paulo at gmail.com (Paulo Vicentini) Date: Sat, 18 Dec 2010 12:35:41 -0200 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: References: Message-ID: Yes, realm is necessary, but it is quite "static" and possible to known it before hand Using HA1 is very useful if you wish to prevent storing clear text password of your trunks in the box Paulo On Sat, Dec 18, 2010 at 10:09 AM, Steven Ayre wrote: > Oh ok... yes the a1-hash is in the user directory for people > authenticating to FS... not for FS registering outwards to gateways. > > I remember coming across this in the past and did take a quick look at > how to implement it. > > The main issue I found was that A1 contains the realm, and the realm > is provided by the gateway in the 407 response. You don't therefore > know the realm needed at the time you generate the A1. > > Yes, it would be possible to do by finding the realm the gateway is > using and generating the A1 from that, but if the gateway changes the > realm the A1 will no longer be valid and FS will start failing to > authenticate. > > -Steve > > > On 17 December 2010 21:47, Paulo Vicentini > wrote: > > Hi Steve > > Yes, it is all about a1-hash > > But I did not see support for storing HA1 for a gateway (UAC), even in > > sofia-lib > > So that both sofia-lib and freeswitch would need to be patched for that > aim > > Regards > > Paulo > > > > On Fri, Dec 17, 2010 at 7:29 PM, Steven Ayre > wrote: > >> > >> If you're looking to store passwords encrypted, then that is already > >> supported. > >> > >> Search the Wiki for a1-hash. > >> > >> -Steve > >> > >> > >> > >> On 17 December 2010 21:20, Paulo Vicentini > >> wrote: > >> > Hi, > >> > I would like to override the scheme used for digest authorization > >> > Actually it is using the scheme coming from sip_www_authenticate_t in > >> > the > >> > sofia_reg_handle_sip_r_challenge function, ignoring scheme set in xml > >> > configuration > >> > > >> > I would like something like: > >> > if(gateway->register_scheme) > >> > scheme = gateway->register_scheme; > >> > before nua_authenticate(... > >> > I am patching sofia to accept HA1 > >> > Regards > >> > Paulo > >> > > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-dev mailing list > >> > FreeSWITCH-dev at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-dev > >> > http://www.freeswitch.org > >> > > >> > > >> > >> _______________________________________________ > >> FreeSWITCH-dev mailing list > >> FreeSWITCH-dev at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > >> http://www.freeswitch.org > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101218/52ef2acd/attachment.html From brian at freeswitch.org Sat Dec 18 19:15:12 2010 From: brian at freeswitch.org (Brian West) Date: Sat, 18 Dec 2010 10:15:12 -0600 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: References: Message-ID: <2B806E21-B705-4CA9-8981-FB1134466FBC@freeswitch.org> the A1 hash is NO more secure if they have that its the exact same as having the username and password anyway. /b On Dec 18, 2010, at 8:35 AM, Paulo Vicentini wrote: > Yes, realm is necessary, but it is quite "static" and possible to known it before hand > Using HA1 is very useful if you wish to prevent storing clear text password of your trunks in the box > > Paulo > From vicentini.paulo at gmail.com Sat Dec 18 20:13:09 2010 From: vicentini.paulo at gmail.com (Paulo Vicentini) Date: Sat, 18 Dec 2010 15:13:09 -0200 Subject: [Freeswitch-dev] overriding authentication scheme on mod_sofia In-Reply-To: <2B806E21-B705-4CA9-8981-FB1134466FBC@freeswitch.org> References: <2B806E21-B705-4CA9-8981-FB1134466FBC@freeswitch.org> Message-ID: Yes...indeed but some applications store in their databases HA1 rather than clear text username and password /r Paulo On Sat, Dec 18, 2010 at 2:15 PM, Brian West wrote: > the A1 hash is NO more secure if they have that its the exact same as > having the username and password anyway. > > /b > > On Dec 18, 2010, at 8:35 AM, Paulo Vicentini wrote: > > > Yes, realm is necessary, but it is quite "static" and possible to known > it before hand > > Using HA1 is very useful if you wish to prevent storing clear text > password of your trunks in the box > > > > Paulo > > > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101218/c63de850/attachment.html From chad at apartmentlines.com Sun Dec 19 05:46:34 2010 From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines) Date: Sat, 18 Dec 2010 18:46:34 -0800 Subject: [Freeswitch-dev] Jester pre-alpha released Message-ID: Jester, the new Lua scripting toolkit for FreeSWITCH, is now officially released in the pre-alpha phase. Code is available in the freeswitch-contrib repository under hunmonk/jester. To learn more, you can start here: http://wiki.freeswitch.org/wiki/Jester -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101218/36d8892d/attachment.html From mike at jerris.com Mon Dec 20 13:58:33 2010 From: mike at jerris.com (Michael Jerris) Date: Mon, 20 Dec 2010 05:58:33 -0500 Subject: [Freeswitch-dev] Freeswitch & Solaris 10 In-Reply-To: <9A8DB267-9125-4C73-9A4B-519BF6AEBDC1@maui-systems.co.uk> References: <9A8DB267-9125-4C73-9A4B-519BF6AEBDC1@maui-systems.co.uk> Message-ID: This is already on my short list to fix. Keep an eye out for a commit soon. Mike On Dec 14, 2010, at 2:22 PM, Marty Lee wrote: > > Hi, > > just playing with building FreeSwitch on a Solaris 10 box and am hitting a link > error when building the target '.libs/freeswitch' > > Undefined first referenced > symbol in file > make_mask32 freeswitch-switch.o > ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch > > Looking at the source, make_mask32 seems to be defined in the spandsp/src/bit_operations.c file, which isn't involved in the linking at > this point. > > Building with Sun Studio 12, just in case it's of any interest. > > I'll try and move the make_mask32 function in to the library and see if that cures > the problem. Thought I'd ask here, as this kind of thing should appear on most > platforms if it really is a problem - meaning it might be something at my end. > > m > > > ----- > Marty Lee e: marty at maui-systems.co.uk > Technical Director v: +44 845 869 2661 > Maui Systems Ltd f: +44 871 433 8922 > Scotland, UK w: http://www.maui-systems.co.uk > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org From bggoutham at gmail.com Mon Dec 20 18:46:31 2010 From: bggoutham at gmail.com (Goutham BG) Date: Mon, 20 Dec 2010 21:16:31 +0530 Subject: [Freeswitch-dev] Query related to enabling SRTP in FreeSWITCH-1.0.7 Message-ID: Hi, I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues. I have the following entry in my dialplan XML file: * * A SIP phone (Avaya 12XX) configured in "SRTP best effort" mode dials into this extension and is connected to the IVR. But the media is established in SRTP in one way and RTP in the other way. The phone offers the following SDP in the INVITE message: v=0 o=- 10170 10170 IN IP4 47.152.232.147 s=Sip Call c=IN IP4 47.152.232.147 t=0 0 m=audio 5016 RTP/AVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=sendrecv m=audio 5016 RTP/SAVP 0 8 18 101 102 a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=ptime:20 a=rtpmap:18 G729/8000 a=ptime:20 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 X-nt-inforeq/8000 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Tjivoci1I/mVkt/Fq/ZsiY+ +ornJoXjZ5tSadho4 a=sendrecv As we can see, there are two "m=" lines in the SDP of the offer; one for RTP and another for SRTP. FreeSWITCH-1.0.7 answers the call by sending 200OK with the following SDP: v=0 o=FreeSWITCH 1291628984 1291628985 IN IP4 47.152.232.156 s=FreeSWITCH c=IN IP4 47.152.232.156 t=0 0 m=audio 11280 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=audio 0 RTP/SAVP 19 As you can see above, FreeSWITCH accepts the RTP stream and rejects the SRTP stream (by sending port as 0) in the SDP. The SIP phone sends the media in RTP(which is expected). But, FreeSWITCH sends the media in SRTP to the SIP phone. I believe this is a bug in FreeSWITCH as it is supposed to send the media in RTP since it accepted RTP in the answer (200OK). *Query: ======* In order to make FreeSWITCH select SRTP in the SDP of the answer(200OK), I made the following change(in *bold*) in FS dial plan: * * In FreeSWITCH-1.0.6(before updating to 1.0.7), this worked and FS accepted the SRTP stream and rejected RTP in the answer(200 OK) as shown below: m=audio 0 RTP/AVP 19 m=audio 12084 RTP/SAVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:hgv7ClqDx1irTRrXq2NEm9Gbouw0969bBU3n+LcM But after updating the FreeSWITCH-1.0.6 to 1.0.7, the above mentioned dial plan change (i.e, setting sip_secure_media=true) is not working. It is still behaving in the same way as it did without the XML change. Can you please let me know if anything else needs to be added in dialplan XML file for enabling SRTP in this case in FreeSWITCH-1.0.7 or am I missing something here? I have referred the following FS wiki pages for making the SRTP changes: http://wiki.freeswitch.org/wiki/Secure_RTP http://wiki.freeswitch.org/wiki/SRTP Note: There is no issue when the SIP phone is configured in "SRTP only" mode where only SRTP stream is offered in the SDP of the INVITE. In this case, SIP phone and FreeSWITCH communicate properly using SRTP. This doesn't require setting "sip_secure_media=true" in the dialplan XML file. P.S: I am a newbie to FreeSWITCH. So, please forgive me if I am asking basic questions. Thanks Goutham B G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101220/6572b7eb/attachment.html From brian at freeswitch.org Mon Dec 20 19:11:36 2010 From: brian at freeswitch.org (Brian West) Date: Mon, 20 Dec 2010 10:11:36 -0600 Subject: [Freeswitch-dev] Query related to enabling SRTP in FreeSWITCH-1.0.7 In-Reply-To: References: Message-ID: <6B5ED5A8-0012-4575-9530-11298F498CDD@freeswitch.org> You have to always set this before you answer the call. And its behavior has NOT changed. /b On Dec 20, 2010, at 9:46 AM, Goutham BG wrote: > Hi, > > I have been trying to enable SRTP in FreeSWITCH-1.0.7 and have been facing some issues. > I have the following entry in my dialplan XML file: From jmesquita at freeswitch.org Sun Dec 26 04:22:13 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 25 Dec 2010 22:22:13 -0300 Subject: [Freeswitch-dev] mod_conference tracking Message-ID: Hello you all. I hope you are all having a good holiday. I have been looking for the best way of tracking conference usage for reporting purposes (basically answering the question: what happened and when on conference n? X?) and I thought of the following alternatives that I would like to discuss with you. 1. Using ESL and the CUSTOM events to keep track of what happened when (I guess that's the most widely used) 2. Creating a new entry on switch_caller_profile, that would be filled by the mod_conference module anytime something changes on the conference. Something similar to what the origination_caller_profile does but that would be manipulated by the mod_conference. This option could neatly integrate with the xml cdrs but would mess a little more on the core. Not sure it is desirable. 3. Make mod_conference spit it's own XML for each conference created so that we know what happens and can link offline to other cdr entities. Or, maybe I am completely dumb and trying to do something that it is not supposed to be done or forgetting some resource that is already available. Jo?o Mesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101225/c5fbf1d0/attachment.html From jmesquita at freeswitch.org Mon Dec 27 07:31:59 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 27 Dec 2010 01:31:59 -0300 Subject: [Freeswitch-dev] mod_conference tracking In-Reply-To: References: Message-ID: I decided, for now, to take the short way out. That is, use ESL to track the events on the conference. One problem that emerged while doing so was how to find unique instances of a certain conference. To make it clearer, let me make an example. Conference 123 is created on Sep 20, 2011 and has 20 members. Same conference 123 is created on Sep 21, 2011 but now with 10 members. Besides the obvious time difference, there was a problem identifying that the conferences are unique in memory in FreeSWITCH, more accurately, the same conference_obj. I came up with the idea then of adding one member to the struct. That member is a uuid for that conference and it is added to all events spit out by the module, so we know which conference is which uniquely. Since this patch adds stuff to the struct, I didn't want to just commit without approval, so I would like to kindly ask one of the core devs to take a look at it and give me the green light to commit it, if that's the case. Sorry for posting it here, but Jira has been out for most of the weekend. I will get it in there if it gets back online and this email has not been responded yet. The patch is attached. Thank you, Jo?o Mesquita 2010/12/25 Jo?o Mesquita > Hello you all. I hope you are all having a good holiday. > > I have been looking for the best way of tracking conference usage for > reporting purposes (basically answering the question: what happened and when > on conference n? X?) and I thought of the following alternatives that I > would like to discuss with you. > > 1. Using ESL and the CUSTOM events to keep track of what happened when (I > guess that's the most widely used) > > 2. Creating a new entry on switch_caller_profile, that would be filled by > the mod_conference module anytime something changes on the conference. > Something similar to what the origination_caller_profile does but that would > be manipulated by the mod_conference. This option could neatly integrate > with the xml cdrs but would mess a little more on the core. Not sure it is > desirable. > > 3. Make mod_conference spit it's own XML for each conference created so > that we know what happens and can link offline to other cdr entities. > > Or, maybe I am completely dumb and trying to do something that it is not > supposed to be done or forgetting some resource that is already available. > > > Jo?o Mesquita > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101227/7ed08807/attachment.html -------------- next part -------------- diff --git a/src/mod/applications/mod_conference/mod_conference.c b/src/mod/applications/mod_conference/mod_conference.c index fa1f12c..22d3b40 100644 --- a/src/mod/applications/mod_conference/mod_conference.c +++ b/src/mod/applications/mod_conference/mod_conference.c @@ -287,6 +287,7 @@ typedef struct conference_obj { uint32_t avg_itt; uint32_t avg_tally; switch_time_t run_time; + switch_uuid_t uuid; } conference_obj_t; /* Relationship with another member */ @@ -435,11 +436,14 @@ static switch_status_t conference_add_event_member_data(conference_member_t *mem static switch_status_t conference_add_event_data(conference_obj_t *conference, switch_event_t *event) { + char uuid_str[SWITCH_UUID_FORMATTED_LENGTH+1]; switch_status_t status = SWITCH_STATUS_SUCCESS; + switch_uuid_format(uuid_str, &(conference->uuid)); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Conference-Name", conference->name); switch_event_add_header(event, SWITCH_STACK_BOTTOM, "Conference-Size", "%u", conference->count); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Conference-Profile-Name", conference->profile_name); + switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Conference-Unique-ID", uuid_str); return status; } @@ -2779,6 +2783,12 @@ static void *SWITCH_THREAD_FUNC conference_record_thread_run(switch_thread_t *th switch_core_file_close(&fh); } switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "Recording of %s Stopped\n", rec->path); + if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM, CONF_EVENT_MAINT) == SWITCH_STATUS_SUCCESS) { + conference_add_event_data(conference, event); + switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Action", "stop-recording"); + switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Path", rec->path); + switch_event_fire(&event); + } if (rec->pool) { switch_memory_pool_t *pool = rec->pool; @@ -6315,6 +6325,9 @@ static conference_obj_t *conference_new(char *name, conf_xml_cfg_t cfg, switch_m conference->verbose_events = 1; } + /* Create the conference unique identifier */ + switch_uuid_get(&(conference->uuid)); + /* Activate the conference mutex for exclusivity */ switch_mutex_init(&conference->mutex, SWITCH_MUTEX_NESTED, conference->pool); switch_mutex_init(&conference->flag_mutex, SWITCH_MUTEX_NESTED, conference->pool); From anthony.minessale at gmail.com Mon Dec 27 18:45:02 2010 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 27 Dec 2010 09:45:02 -0600 Subject: [Freeswitch-dev] mod_conference tracking In-Reply-To: References: Message-ID: you might want to consider putting the uuid_str in the conference struct and declare the uuid_t local and just render it once into the string field. Then you never have to call the uuid_format again and you probably don't need the binary version for anything. Also do you add this field to every event in the general function that prepares all conference events for sending? 2010/12/26 Jo?o Mesquita : > I decided, for now, to take the short way out. That is, use ESL to track the > events on the conference. > > One problem that emerged while doing so was how to find unique instances of > a certain conference. To make it clearer, let me make an example. > > Conference 123 is created on Sep 20, 2011 and has 20 members. > > Same conference 123 is created on Sep 21, 2011 but now with 10 members. > > Besides the obvious time difference, there was a problem identifying that > the conferences are unique in memory in FreeSWITCH, more accurately, the > same conference_obj. I came up with the idea then of adding one member to > the struct. That member is a uuid for that conference and it is added to all > events spit out by the module, so we know which conference is which > uniquely. > > Since this patch adds stuff to the struct, I didn't want to just commit > without approval, so I would like to kindly ask one of the core devs to take > a look at it and give me the green light to commit it, if that's the case. > Sorry for posting it here, but Jira has been out for most of the weekend. I > will get it in there if it gets back online and this email has not been > responded yet. > > The patch is attached. > > Thank you, > Jo?o Mesquita > > > 2010/12/25 Jo?o Mesquita >> >> Hello you all. I hope you are all having a good holiday. >> I have been looking for the best way of tracking conference usage for >> reporting purposes (basically answering the question: what happened and when >> on conference n? X?) and I thought of the following alternatives that I >> would like to discuss with you. >> 1. Using ESL and the CUSTOM events to keep track of what happened when (I >> guess that's the most widely used) >> 2. Creating a new entry on switch_caller_profile, that would be filled by >> the mod_conference module anytime something changes on the conference. >> Something similar to what the origination_caller_profile does but that would >> be manipulated by the mod_conference. This option could neatly integrate >> with the xml cdrs but would mess a little more on the core. Not sure it is >> desirable. >> 3. Make mod_conference spit it's own XML for each conference created so >> that we know what happens and can link offline to other cdr entities. >> Or, maybe I am completely dumb and trying to do something that it is not >> supposed to be done or forgetting some resource that is already available. >> >> Jo?o Mesquita > > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jmesquita at freeswitch.org Tue Dec 28 03:31:26 2010 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 27 Dec 2010 21:31:26 -0300 Subject: [Freeswitch-dev] mod_conference tracking In-Reply-To: References: Message-ID: I made the modifications as you suggested, tested under latest git and committed. Yes, I add this field to every event since we already had conference_add_event_data in place. Thank you. Regards, Jo?o Mesquita On Mon, Dec 27, 2010 at 12:45 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you might want to consider putting the uuid_str in the conference > struct and declare the uuid_t local and just > render it once into the string field. Then you never have to call the > uuid_format again and you probably don't need the binary version for > anything. > > Also do you add this field to every event in the general function that > prepares all conference events for sending? > > > 2010/12/26 Jo?o Mesquita : > > I decided, for now, to take the short way out. That is, use ESL to track > the > > events on the conference. > > > > One problem that emerged while doing so was how to find unique instances > of > > a certain conference. To make it clearer, let me make an example. > > > > Conference 123 is created on Sep 20, 2011 and has 20 members. > > > > Same conference 123 is created on Sep 21, 2011 but now with 10 members. > > > > Besides the obvious time difference, there was a problem identifying that > > the conferences are unique in memory in FreeSWITCH, more accurately, the > > same conference_obj. I came up with the idea then of adding one member to > > the struct. That member is a uuid for that conference and it is added to > all > > events spit out by the module, so we know which conference is which > > uniquely. > > > > Since this patch adds stuff to the struct, I didn't want to just commit > > without approval, so I would like to kindly ask one of the core devs to > take > > a look at it and give me the green light to commit it, if that's the > case. > > Sorry for posting it here, but Jira has been out for most of the weekend. > I > > will get it in there if it gets back online and this email has not been > > responded yet. > > > > The patch is attached. > > > > Thank you, > > Jo?o Mesquita > > > > > > 2010/12/25 Jo?o Mesquita > >> > >> Hello you all. I hope you are all having a good holiday. > >> I have been looking for the best way of tracking conference usage for > >> reporting purposes (basically answering the question: what happened and > when > >> on conference n? X?) and I thought of the following alternatives that I > >> would like to discuss with you. > >> 1. Using ESL and the CUSTOM events to keep track of what happened when > (I > >> guess that's the most widely used) > >> 2. Creating a new entry on switch_caller_profile, that would be filled > by > >> the mod_conference module anytime something changes on the conference. > >> Something similar to what the origination_caller_profile does but that > would > >> be manipulated by the mod_conference. This option could neatly integrate > >> with the xml cdrs but would mess a little more on the core. Not sure it > is > >> desirable. > >> 3. Make mod_conference spit it's own XML for each conference created so > >> that we know what happens and can link offline to other cdr entities. > >> Or, maybe I am completely dumb and trying to do something that it is not > >> supposed to be done or forgetting some resource that is already > available. > >> > >> Jo?o Mesquita > > > > > > _______________________________________________ > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101227/119f37e0/attachment.html From msc at freeswitch.org Wed Dec 29 20:06:35 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Dec 2010 09:06:35 -0800 Subject: [Freeswitch-dev] FreeSWITCH Conference Call Today Message-ID: Hey gang, Here is today's agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2010_12_29 It is light since we have lots of people on vacation and otherwise taking holiday. If everyone is up to it I'd like to do what we did last week and have everyone help out with some documentation. We have several things that we can work on: New jitterbuffer API FIFO params and vars Voicemail page needs to be updated Thanks! Talk to you shortly. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101229/1a05ffae/attachment.html From msc at freeswitch.org Thu Dec 30 03:16:44 2010 From: msc at freeswitch.org (Michael Collins) Date: Wed, 29 Dec 2010 16:16:44 -0800 Subject: [Freeswitch-dev] New FS sounds testing Message-ID: FreeSWITCHers, We have the new sounds rolled and I'd like to do some expanded testing before I update in git. If anyone wants to get the latest sounds then go to your src directory and edit build/sounds_version.txt. Change en-us-callie from 1.0.13 to 1.0.14 and then do "make cd-sounds-install". The latest sounds should get installed. Please try this on your non-production systems first! :) Please reply to me off-list and let me know if you run into any problems or if the sounds are okay. I would like feedback either way. If we don't have any issues then I'll bump the sounds version in git and everyone will be good to go. Thanks! -Michael P.S. - If you would like to review which sound prompts are new to 1.0.14 then check this commit: http://fisheye.freeswitch.org/browse/freeswitch.git/docs/phrase/phrase_en.xml?r1=ee051faef39ff78d7805f1eb8c9f6330a4258032&r2=257c7edaf7d7e6151d11e5ef924d87a77f2c369b -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20101229/a68d84cc/attachment.html