From dujinfang at gmail.com Sat Aug 1 08:10:31 2009 From: dujinfang at gmail.com (Seven Du) Date: Sat, 1 Aug 2009 23:10:31 +0800 Subject: [Freeswitch-dev] what's the best way to change code and make patch In-Reply-To: <191c3a030907312200x517c27bdse0b39ad950778ef8@mail.gmail.com> References: <77099A23-482F-40E4-9957-0C35F0C1F39F@freeswitch.org> <353D9FB8-A25A-438E-80B1-3B347F132C36@gmail.com> <36F09EE7-8836-4FD2-A248-3A95577C1CB2@freeswitch.org> <62BD09A8-D64A-4D90-8BED-738B764E18B4@jerris.com> <191c3a030907312200x517c27bdse0b39ad950778ef8@mail.gmail.com> Message-ID: On Aug 1, 2009, at 1:00 PM, Anthony Minessale wrote: > dont forget to propose what you are patching so we all approve of it > if you plan to re-submit. > Thank you Anthony, I'd like to ask this question even no patch :) Just like this one: http://jira.freeswitch.org/browse/MODENDP-236 And I'm thinking to add a little more statistic messages to profile and gateways. While profile has failed calls stat, gateways don't. And something like number of failed calls in the last 5 minutes or number of continuously failed calls, and might be fired to event socket to do smart routings. Sure it's easy to listen to the event socket and do whatever we want, some features are better to be built in. Anyone think we need a built in statistic Mod to generate reports on resource usage statictics? Another one would be api to remove a fifo node from memory. OK, another question, if, just if I have code in branch, can it be directly merged to trunk without a jira? Thank you. > On Fri, Jul 31, 2009 at 10:34 PM, Michael Jerris > wrote: > If you use git locally with git-svn you can do this. > > git add -i > Thank you Michael I think that works. From gmaruzz at celliax.org Wed Aug 5 14:00:17 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Wed, 5 Aug 2009 23:00:17 +0200 Subject: [Freeswitch-dev] freepbx for freeswitch Message-ID: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com> Yay! http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future Darren Schreiber has made the announcement and is doinng a presentation of FreePBX V3 right now at www.cluecon.com. From jmesquita at gmail.com Wed Aug 5 22:06:27 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 6 Aug 2009 02:06:27 -0300 Subject: [Freeswitch-dev] ESL UUIDs Message-ID: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> What is the difference between a Core-UUID and Unique-ID? Wiki page states that all events should have a Core-UUID, but when log 7 sends events that do not have a Core-UUID. I am looking for something that can be a hash key to all events sent from a server. I don't have a problem using Unique-ID, but that is not present on all events (obviously). jmesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090806/92db90ec/attachment.html From mrene_lists at avgs.ca Wed Aug 5 22:34:47 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 6 Aug 2009 00:34:47 -0500 Subject: [Freeswitch-dev] ESL UUIDs In-Reply-To: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> References: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> Message-ID: <1D0741BC-9D8F-44C4-B0DF-0DF6BDCBD24A@avgs.ca> Core-UUID identifies the FS instance ID. Which events dont include it? we can fix that Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca Am 6-Aug-09 um 12:06 AM schrieb Jo?o Mesquita: > What is the difference between a Core-UUID and Unique-ID? > > Wiki page states that all events should have a Core-UUID, but when > log 7 sends events that do not have a Core-UUID. > > I am looking for something that can be a hash key to all events sent > from a server. I don't have a problem using Unique-ID, but that is > not present on all events (obviously). > > jmesquita > _______________________________________________ > 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 anthony.minessale at gmail.com Thu Aug 6 05:50:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 6 Aug 2009 07:50:55 -0500 Subject: [Freeswitch-dev] ESL UUIDs In-Reply-To: <1D0741BC-9D8F-44C4-B0DF-0DF6BDCBD24A@avgs.ca> References: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> <1D0741BC-9D8F-44C4-B0DF-0DF6BDCBD24A@avgs.ca> Message-ID: <191c3a030908060550q185cd100vcfaa3c162448b5a9@mail.gmail.com> The logging events do not currently send the core-uuid but it would not be much of a hassle to add it. On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene wrote: > Core-UUID identifies the FS instance ID. Which events dont include it? > we can fix that > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > Am 6-Aug-09 um 12:06 AM schrieb Jo?o Mesquita: > > > What is the difference between a Core-UUID and Unique-ID? > > > > Wiki page states that all events should have a Core-UUID, but when > > log 7 sends events that do not have a Core-UUID. > > > > I am looking for something that can be a hash key to all events sent > > from a server. I don't have a problem using Unique-ID, but that is > > not present on all events (obviously). > > > > jmesquita > > _______________________________________________ > > 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 > -- 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090806/5ec98782/attachment-0001.html From jmesquita at gmail.com Thu Aug 6 06:48:58 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 6 Aug 2009 10:48:58 -0300 Subject: [Freeswitch-dev] ESL UUIDs In-Reply-To: <191c3a030908060550q185cd100vcfaa3c162448b5a9@mail.gmail.com> References: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> <1D0741BC-9D8F-44C4-B0DF-0DF6BDCBD24A@avgs.ca> <191c3a030908060550q185cd100vcfaa3c162448b5a9@mail.gmail.com> Message-ID: <5a8712120908060648m3dcd6415udaf76dd87b371507@mail.gmail.com> I was thinking of adding a new macro to also add Unique-ID to the log messages that have sessions associated with it. I have talked to MikeJ a bit about it and it would be good for FsGui cos I would be able to trace specific calls on a huge log. Can you guys point me into the direction of getting this done? I know it is a lot of work because it involves modifying lots of macros. jmesquita On Thu, Aug 6, 2009 at 9:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The logging events do not currently send the core-uuid but it would not be > much of a hassle to add it. > > > > On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene wrote: > >> Core-UUID identifies the FS instance ID. Which events dont include it? >> we can fix that >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> Am 6-Aug-09 um 12:06 AM schrieb Jo?o Mesquita: >> >> > What is the difference between a Core-UUID and Unique-ID? >> > >> > Wiki page states that all events should have a Core-UUID, but when >> > log 7 sends events that do not have a Core-UUID. >> > >> > I am looking for something that can be a hash key to all events sent >> > from a server. I don't have a problem using Unique-ID, but that is >> > not present on all events (obviously). >> > >> > jmesquita >> > _______________________________________________ >> > 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 >> > > > > -- > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > 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/20090806/60e6fed5/attachment.html From jmesquita at gmail.com Thu Aug 6 06:50:50 2009 From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 6 Aug 2009 10:50:50 -0300 Subject: [Freeswitch-dev] ESL UUIDs In-Reply-To: <191c3a030908060550q185cd100vcfaa3c162448b5a9@mail.gmail.com> References: <5a8712120908052206v4b7dcc0ds5d8f1910c4d0d643@mail.gmail.com> <1D0741BC-9D8F-44C4-B0DF-0DF6BDCBD24A@avgs.ca> <191c3a030908060550q185cd100vcfaa3c162448b5a9@mail.gmail.com> Message-ID: <5a8712120908060650k51778162ma4be3df860412fdc@mail.gmail.com> Forgot to say that Core-UUID is really what I need cos my data structure uses that to separate events from each FS instance. It is not extremely necessary, but it would be neat to use and less error prone. If I understood it correctly, Core-UUID never changes for each FS instance. Is that right? jmesquita On Thu, Aug 6, 2009 at 9:50 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The logging events do not currently send the core-uuid but it would not be > much of a hassle to add it. > > > > On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene wrote: > >> Core-UUID identifies the FS instance ID. Which events dont include it? >> we can fix that >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> Am 6-Aug-09 um 12:06 AM schrieb Jo?o Mesquita: >> >> > What is the difference between a Core-UUID and Unique-ID? >> > >> > Wiki page states that all events should have a Core-UUID, but when >> > log 7 sends events that do not have a Core-UUID. >> > >> > I am looking for something that can be a hash key to all events sent >> > from a server. I don't have a problem using Unique-ID, but that is >> > not present on all events (obviously). >> > >> > jmesquita >> > _______________________________________________ >> > 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 >> > > > > -- > 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 > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > 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/20090806/5249dfbf/attachment.html From msc at freeswitch.org Thu Aug 6 10:30:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 6 Aug 2009 12:30:09 -0500 Subject: [Freeswitch-dev] FreeSWITCH 1.0.4 Release Announcement Message-ID: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com> We are happy to announce the official release of FreeSWITCH 1.0.4! Please visit this link to Digg and read the story, and then spread the word! Thanks for being such a great community! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090806/c1d57207/attachment.html From dome at tel.co.th Thu Aug 6 21:07:00 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 7 Aug 2009 11:07:00 +0700 Subject: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement In-Reply-To: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com> References: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com> Message-ID: <8ccbff060908062107p1f21fd8dp8c631d7a627520f8@mail.gmail.com> Good News.. 2009/8/7 Michael Collins : > We are happy to announce the official release of FreeSWITCH 1.0.4! Please > visit this link to Digg and read the story, and then spread the word! > > Thanks for being such a great community! > -Michael > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From cooper.xu at mapleworks.com Thu Aug 6 14:29:34 2009 From: cooper.xu at mapleworks.com (Cooper Xu) Date: Thu, 6 Aug 2009 17:29:34 -0400 Subject: [Freeswitch-dev] Problem with RTP auto_flush code and hot_socket Message-ID: <6F56CF55DCE641E28BF25184C8DD7B38@mapleworks.com> Hi, In our load test, we found that the new RTP auto_flush and hot_socket code after version 1.04 seems to cause voice quality problem. We were stress testing Freeswitch with 250 channels using codec G.711u and iLBC. The RTP auto_flush appeared to be trigged very frequently. Because our system is under load and socket buffer for RTP stream sometimes accumulated a few packets before the channel can handle it. When this happened, the RTP auto_flush and hot_socket mechanism seemed sending those accumulated packets in a very short interval. However after this burst, the RTP stream receiving will wait for 80-100ms before receiving and sending another RTP packet. To find out why this can happen, We looked at the RTP part of Freeswitch code and found that inside the RTP read, during auto RTP flush, it will call switch_core_timer_sync() function. This will add 3 to the timer reference count and the next switch_core_timer_next() call will add another 1 to timer reference. This caused the next RTP receiving being delayed 80-100ms. This 80-100ms delay for RTP receiving caused 4-5 packets being accumulated in socket buffer again. And it will soon trigger another auto_flush for RTP stream. As the result, we saw a lot of short burst in the RTP stream, which caused voice quality problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090806/24fa60cd/attachment-0001.html From brian at freeswitch.org Sat Aug 8 07:08:14 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 8 Aug 2009 09:08:14 -0500 Subject: [Freeswitch-dev] Problem with RTP auto_flush code and hot_socket In-Reply-To: <6F56CF55DCE641E28BF25184C8DD7B38@mapleworks.com> References: <6F56CF55DCE641E28BF25184C8DD7B38@mapleworks.com> Message-ID: Please Open a jira... this is the second report I have heard of this issue. /b On Aug 6, 2009, at 4:29 PM, Cooper Xu wrote: > Hi, > > In our load test, we found that the new RTP auto_flush and > hot_socket code after version 1.04 seems to cause voice quality > problem. We were stress testing Freeswitch with 250 channels using > codec G.711u and iLBC. The RTP auto_flush appeared to be trigged > very frequently. > > Because our system is under load and socket buffer for RTP stream > sometimes accumulated a few packets before the channel can handle > it. When this happened, the RTP auto_flush and hot_socket mechanism > seemed sending those accumulated packets in a very short interval. > However after this burst, the RTP stream receiving will wait for > 80-100ms before receiving and sending another RTP packet. To find > out why this can happen, We looked at the RTP part of Freeswitch > code and found that inside the RTP read, during auto RTP flush, it > will call switch_core_timer_sync() function. This will add 3 to the > timer reference count and the next switch_core_timer_next() call > will add another 1 to timer reference. This caused the next RTP > receiving being delayed 80-100ms. This 80-100ms delay for RTP > receiving caused 4-5 packets being accumulated in socket buffer > again. And it will soon trigger another auto_flush for RTP stream. > As the result, we saw a lot of short burst in the RTP stream, which > caused voice quality problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090808/ac6b5e51/attachment.html From freeswitch-list at puzzled.xs4all.nl Sun Aug 9 12:18:56 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Sun, 09 Aug 2009 21:18:56 +0200 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) Message-ID: <4A7F2120.3030802@puzzled.xs4all.nl> Hi, I was reworking the freeswitch.spec file to make it build an RPM on Fedora 11 and noticed the following: [..snip a gazillion previous ones..] ERROR 0002: file '/opt/freeswitch/mod/mod_sndfile.so' contains an invalid rpath '/opt/freeswitch/mod' in [/opt/freeswitch/lib64:/opt/freeswitch/mod] [..snip a gazillion more..] To solve this issue I had to add in the spec file after configure the following lines: # kill rpath madness in the top level libtool sed -i 's|^hardcode_libdir_flag_spec=.*|hardcode_libdir_flag_spec=""|g' libtool sed -i 's|^runpath_var=LD_RUN_PATH|runpath_var=DIE_RPATH_DIE|g' libtool # and fix it in the libtool in libs/openzap too sed -i 's|^hardcode_libdir_flag_spec=.*|hardcode_libdir_flag_spec=""|g' libs/openzap/libtool sed -i 's|^runpath_var=LD_RUN_PATH|runpath_var=DIE_RPATH_DIE|g' libs/openzap/libtool Maybe this could be done on libtool in trunk? If I need to report this in Jira please let me know. Regards, Patrick From mike at jerris.com Sun Aug 9 21:35:47 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Aug 2009 00:35:47 -0400 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <4A7F2120.3030802@puzzled.xs4all.nl> References: <4A7F2120.3030802@puzzled.xs4all.nl> Message-ID: <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> While that mod doesn't need that in the rpath, some others do. What exactly is invalid about it? On Aug 9, 2009, at 3:18 PM, Patrick wrote: > Hi, > > I was reworking the freeswitch.spec file to make it build an RPM on > Fedora 11 and noticed the following: > > [..snip a gazillion previous ones..] > > ERROR 0002: file '/opt/freeswitch/mod/mod_sndfile.so' contains an > invalid rpath '/opt/freeswitch/mod' in > [/opt/freeswitch/lib64:/opt/freeswitch/mod] > > [..snip a gazillion more..] > > To solve this issue I had to add in the spec file after configure the > following lines: > > # kill rpath madness in the top level libtool > sed -i 's|^hardcode_libdir_flag_spec=.*|hardcode_libdir_flag_spec=""| > g' > libtool > sed -i 's|^runpath_var=LD_RUN_PATH|runpath_var=DIE_RPATH_DIE|g' > libtool > > # and fix it in the libtool in libs/openzap too > sed -i 's|^hardcode_libdir_flag_spec=.*|hardcode_libdir_flag_spec=""| > g' > libs/openzap/libtool > sed -i 's|^runpath_var=LD_RUN_PATH|runpath_var=DIE_RPATH_DIE|g' > libs/openzap/libtool > > Maybe this could be done on libtool in trunk? If I need to report this > in Jira please let me know. > > Regards, > Patrick > > _______________________________________________ > 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 brian at freeswitch.org Sun Aug 9 21:38:36 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 9 Aug 2009 23:38:36 -0500 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <4A7F2120.3030802@puzzled.xs4all.nl> References: <4A7F2120.3030802@puzzled.xs4all.nl> Message-ID: All bugs no matter how minor should go in Jira. Thanks, Brian On Aug 9, 2009, at 2:18 PM, Patrick wrote: > Maybe this could be done on libtool in trunk? If I need to report this > in Jira please let me know. > > Regards, > Patrick From freeswitch-list at puzzled.xs4all.nl Mon Aug 10 09:11:04 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Mon, 10 Aug 2009 18:11:04 +0200 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> References: <4A7F2120.3030802@puzzled.xs4all.nl> <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> Message-ID: <4A804698.30905@puzzled.xs4all.nl> On 08/10/2009 06:35 AM, Michael Jerris wrote: > While that mod doesn't need that in the rpath, some others do. What > exactly is invalid about it? Quoting from http://fedoraproject.org/wiki/Packaging/Guidelines#Beware_of_Rpath "Sometimes, code will hardcode specific library paths when linking binaries (using the -rpath or -R flag). This is commonly referred to as an rpath, and in Fedora it is forbidden." Since Red Hat follows the Fedora Packaging Guidelines this applies to RHEL & CentOS too. Please forgive my ignorance but couldn't a file like /etc/ld.so.conf.d/freeswitch.conf which specifies the libs not solve this? That would work for Fedora, RHEL & CentOS. Regards, Patrick From mike at jerris.com Mon Aug 10 12:08:53 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Aug 2009 12:08:53 -0700 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <4A804698.30905@puzzled.xs4all.nl> References: <4A7F2120.3030802@puzzled.xs4all.nl> <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> <4A804698.30905@puzzled.xs4all.nl> Message-ID: <6F827408-2289-4F29-8E9C-2B271A6816CE@jerris.com> This kills any ability to have multiple versions installed at the same time. On Aug 10, 2009, at 9:11 AM, Patrick wrote: > On 08/10/2009 06:35 AM, Michael Jerris wrote: >> While that mod doesn't need that in the rpath, some others do. What >> exactly is invalid about it? > > Quoting from > http://fedoraproject.org/wiki/Packaging/Guidelines#Beware_of_Rpath > > "Sometimes, code will hardcode specific library paths when linking > binaries (using the -rpath or -R flag). This is commonly referred to > as > an rpath, and in Fedora it is forbidden." > > Since Red Hat follows the Fedora Packaging Guidelines this applies to > RHEL & CentOS too. > > Please forgive my ignorance but couldn't a file like > /etc/ld.so.conf.d/freeswitch.conf which specifies the libs not solve > this? That would work for Fedora, RHEL & CentOS. > > Regards, > Patrick > > _______________________________________________ > 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 freeswitch-list at puzzled.xs4all.nl Mon Aug 10 15:01:10 2009 From: freeswitch-list at puzzled.xs4all.nl (Patrick) Date: Tue, 11 Aug 2009 00:01:10 +0200 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <6F827408-2289-4F29-8E9C-2B271A6816CE@jerris.com> References: <4A7F2120.3030802@puzzled.xs4all.nl> <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> <4A804698.30905@puzzled.xs4all.nl> <6F827408-2289-4F29-8E9C-2B271A6816CE@jerris.com> Message-ID: <4A8098A6.1010503@puzzled.xs4all.nl> On 08/10/2009 09:08 PM, Michael Jerris wrote: > This kills any ability to have multiple versions installed at the same > time. That makes sense. If you know which binaries require rpaths then I can exclude those from the purge in the spec file. Would that be ok? Regards, Patrick From mike at jerris.com Mon Aug 10 18:35:15 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 10 Aug 2009 21:35:15 -0400 Subject: [Freeswitch-dev] Invalid rpaths + fix (1.0.4) In-Reply-To: <4A8098A6.1010503@puzzled.xs4all.nl> References: <4A7F2120.3030802@puzzled.xs4all.nl> <8792939A-BFBF-403E-8AD7-E22E20ED336C@jerris.com> <4A804698.30905@puzzled.xs4all.nl> <6F827408-2289-4F29-8E9C-2B271A6816CE@jerris.com> <4A8098A6.1010503@puzzled.xs4all.nl> Message-ID: <41D665B4-FE29-4DD0-B310-2C71DB14ECA6@jerris.com> Pretty much all of the ones you see the errors on will need rpath, although not the one you see in that error message, but others. Mike On Aug 10, 2009, at 6:01 PM, Patrick wrote: > On 08/10/2009 09:08 PM, Michael Jerris wrote: >> This kills any ability to have multiple versions installed at the >> same >> time. > > That makes sense. If you know which binaries require rpaths then I can > exclude those from the purge in the spec file. Would that be ok? > > Regards, > Patrick > > _______________________________________________ > 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 alexg at etherstack.com Mon Aug 10 19:26:08 2009 From: alexg at etherstack.com (Alex Green) Date: Tue, 11 Aug 2009 12:26:08 +1000 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? Message-ID: <4A80D6C0.8050409@etherstack.com> For FXO I switch back to the dialtone state and collect the digits via the ZAP_SIGEVENT_COLLECTED_DIGIT call back (which is a little bit ugly and does not allow dial ahead, but it works). However, for the E1 PRI card there is no dialtone state, which makes sense as the digits are sent over the D-Channel ('VETO Changing state on 1:1 from UP to DIALTONE' is printed if you try). Openzap must be able to detect digits during a call, but I am having trouble figuring out how. ie. How do you use Openzap to detect in-band DTMF digits during a call? Thanks, -alex From moises.silva at gmail.com Mon Aug 10 20:59:10 2009 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 10 Aug 2009 23:59:10 -0400 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <4A80D6C0.8050409@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> Message-ID: On Mon, Aug 10, 2009 at 10:26 PM, Alex Green wrote: > For FXO I switch back to the dialtone state and collect the digits via > the ZAP_SIGEVENT_COLLECTED_DIGIT call back (which is a little bit ugly > and does not allow dial ahead, but it works). Sounds way too ugly, I don't think you should be doing that. > DIALTONE' is printed if you try). Openzap must be able to detect digits > during a call, but I am having trouble figuring out how. > > ie. How do you use Openzap to detect in-band DTMF digits during a call? > Take a look at mod_openzap/mod_openzap.c, on_clear_channel_signal() It seem to me you should do the same. zap_channel_command(sigmsg->channel, ZAP_COMMAND_ENABLE_DTMF_DETECT, &tt); where tt = ZAP_TONE_DTMF and sigmsg->channel is the zap_channel_t That will enable the DTMF detection feature in the channel (either in software or hardware if its available). After enabling that feature, you can use zap_channel_dequeue_dtmf(zchan, dtmf, sizeof(dtmf)); to dequeue DTMF after calling zap_channel_read(), since there is where dtmf (if any) is enqueued. Check zap_channel_read so you can see how the DTMF is detected and enqueued. If you don't want to call zap_channel_dequeue_dtmf on each read (or you can't), then register an event callback for the span with zap_span_set_event_callback(), then your code will be called with ZAP_EVENT_DTMF when dtmf is received in any channel on that span. Hope that helps, -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090810/6a4c041a/attachment.html From alexg at etherstack.com Mon Aug 10 22:17:58 2009 From: alexg at etherstack.com (Alex Green) Date: Tue, 11 Aug 2009 15:17:58 +1000 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <20090811040854.16C4096C001@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> Message-ID: <4A80FF06.5010208@etherstack.com> Thanks Moises, do I need to do a flush_dtmf() or set values for DTMF on/off period, or anything else? zap_channel_dequeue_dtmf() correctly identifies the digit, but if the digit is held down it is detected more than once. I am using a frame size of 160 for the read. Thanks again, -alex Moises Silva wrote: > On Mon, Aug 10, 2009 at 10:26 PM, Alex Green > wrote: > > For FXO I switch back to the dialtone state and collect the digits via > the ZAP_SIGEVENT_COLLECTED_DIGIT call back (which is a little bit ugly > and does not allow dial ahead, but it works). > > > Sounds way too ugly, I don't think you should be doing that. > > > DIALTONE' is printed if you try). Openzap must be able to detect digits > during a call, but I am having trouble figuring out how. > > ie. How do you use Openzap to detect in-band DTMF digits during a call? > > > Take a look at mod_openzap/mod_openzap.c, on_clear_channel_signal() > > It seem to me you should do the same. > > zap_channel_command(sigmsg->channel, ZAP_COMMAND_ENABLE_DTMF_DETECT, > &tt); where tt = ZAP_TONE_DTMF and sigmsg->channel is the zap_channel_t > > That will enable the DTMF detection feature in the channel (either in > software or hardware if its available). > > After enabling that feature, you can use zap_channel_dequeue_dtmf(zchan, > dtmf, sizeof(dtmf)); to dequeue DTMF after calling zap_channel_read(), > since there is where dtmf (if any) is enqueued. Check zap_channel_read > so you can see how the DTMF is detected and enqueued. > > If you don't want to call zap_channel_dequeue_dtmf on each read (or you > can't), then register an event callback for the span with > zap_span_set_event_callback(), then your code will be called with > ZAP_EVENT_DTMF when dtmf is received in any channel on that span. > > Hope that helps, > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 pawel at voiceworks.pl Tue Aug 11 01:04:11 2009 From: pawel at voiceworks.pl (=?UTF-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Tue, 11 Aug 2009 10:04:11 +0200 Subject: [Freeswitch-dev] bgapi takes 20ms to return job-id Message-ID: <4005AEDE-609C-41F7-A48F-DEA0D7291458@voiceworks.pl> Hi, Any ideas why most calls to bgapi take 20ms (on avg) on my idle quad core FC10 box doing a single transaction at a time ? Can do 300 call per second on that box using SIPP but when a single process tries to control my FS instance using a single ESL socket than 95% of the bgapi calls return job-id in 20ms. The rest takes under a millisecond as it should. Is it possible that thread creation takes 20ms or that the job thread itself gets it's CPU timeslice before the job-id gets sent out (hardly possible with 4 CPU threads) ? I have not done any benchmarking yet to see where the real delay is - but it is visible on the loopback interface and limits the performance of my inbound socket app. a fast one: bgapi uuid_displace fca427df-a8b0-4aff-8041-e328691dea3d start tone_stream://v=-10;%(150,100,400,450);loops=2 ### T +0.001165 127.0.0.1:18022 -> 127.0.0.1:44944 [AP] Content-Type: command/reply Reply-Text: +OK Job-UUID: 73f5ac5f-f3b5-44f0-a839-1b5c785b3ae7 Job-UUID: 73f5ac5f-f3b5-44f0-a839-1b5c785b3ae7 a 20ms delay for similar calls bgapi uuid_displace 380e9f71-f6f2-3b94-3ae0-bc6437bed887 start tone_stream://v=-10;%(150,100,400,450);loops=2 ### T +0.020224 127.0.0.1:18022 -> 127.0.0.1:44944 [AP] Content-Type: command/reply Reply-Text: +OK Job-UUID: a7bc6bc9-b7ed-4c8b-a55d-61a48427ceb9 Job-UUID: a7bc6bc9-b7ed-4c8b-a55d-61a48427ceb9 Pawel, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090811/d8f3e6ef/attachment.html From pawel at voiceworks.pl Tue Aug 11 02:55:46 2009 From: pawel at voiceworks.pl (=?UTF-8?Q?Pawe=C5=82_Pier=C5=9Bcionek?=) Date: Tue, 11 Aug 2009 11:55:46 +0200 Subject: [Freeswitch-dev] bgapi takes 20ms to return job-id In-Reply-To: <4005AEDE-609C-41F7-A48F-DEA0D7291458@voiceworks.pl> References: <4005AEDE-609C-41F7-A48F-DEA0D7291458@voiceworks.pl> Message-ID: <75BBCCF7-FB6C-4F35-958D-66E3ADEE3609@voiceworks.pl> On 2009-08-11, at 10:04, Pawe? Pier?cionek wrote: > Is it possible that thread creation takes 20ms or that the job > thread itself gets it's CPU timeslice before the job-id gets sent > out (hardly possible with 4 CPU threads) ? > Ok, found it. FS waits for the bgapi thread to get it's timeslice in a sanity loop before returning the job-id. Why parse_command waits for it to happen. Even if the thread does not make it in time nothing changes in the reply. This makes most of bgapi commands execute before the requester even gets the job-id back. Synchronous API is in that case opportunistically faster than BGAPI or Async EXEC which are deterministically slow. Pawel, From moises.silva at gmail.com Tue Aug 11 07:08:24 2009 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 11 Aug 2009 10:08:24 -0400 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <4A80FF06.5010208@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> Message-ID: On Tue, Aug 11, 2009 at 1:17 AM, Alex Green wrote: > Thanks Moises, do I need to do a flush_dtmf() or set values for DTMF > on/off period, or anything else? > I think default on/off times are good enough, but I suppose you may have to play with it. As of flushing, sounds like a good idea to do when the call starts, but that depends on your applications, if you are certain of consuming all DTMF always, then is not needed. Are you using hw dtmf? are you using Sangoma boards? I noticed that openzap behavior will be different from an API point of view when using hardware DTMF than when using software DTMF, since the callbacks (for span and channel) are currently not available when using hardware DTMF. That sounds like a bug to me, the behavior should be available regardless of the backend system used for DTMF. I will fix that asap. > zap_channel_dequeue_dtmf() correctly identifies the digit, but if the > digit is held down it is detected more than once. I am using a frame > size of 160 for the read. > More than once like in twice? 3 times, 4 times? ad infinitum if the digit is hold? I'd need to debug to try to figure out where that comes from. If you are using hardware dtmf in wanpipe boards you can go to src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c and search for WP_TDMAPI_EVENT_DTMF, then uncomment the log line there to see if the digit is detected twice by the drivers. If using software DTMF, some logging in zap_channel_read when calling teletone_dtmf_detect() can be added to check if its detecting it twice. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090811/95c366d1/attachment.html From mrene_lists at avgs.ca Tue Aug 11 08:16:00 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 11 Aug 2009 11:16:00 -0400 Subject: [Freeswitch-dev] bgapi takes 20ms to return job-id In-Reply-To: <75BBCCF7-FB6C-4F35-958D-66E3ADEE3609@voiceworks.pl> References: <4005AEDE-609C-41F7-A48F-DEA0D7291458@voiceworks.pl> <75BBCCF7-FB6C-4F35-958D-66E3ADEE3609@voiceworks.pl> Message-ID: I checked the commit log and it was added by anthm to fix a race condition, I am not sure on the details though. However, the bgapi api command doesn't have anything like that, you might want to try sending api bgapi .... and see if you get the same problem, it should return right away. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 11-Aug-09, at 5:55 AM, Pawe? Pier?cionek wrote: > > On 2009-08-11, at 10:04, Pawe? Pier?cionek wrote: > >> Is it possible that thread creation takes 20ms or that the job >> thread itself gets it's CPU timeslice before the job-id gets sent >> out (hardly possible with 4 CPU threads) ? >> > > Ok, found it. FS waits for the bgapi thread to get it's timeslice in a > sanity loop before returning the job-id. > Why parse_command waits for it to happen. Even if the thread does not > make it in time nothing changes in the reply. > This makes most of bgapi commands execute before the requester even > gets the job-id back. > Synchronous API is in that case opportunistically faster than BGAPI or > Async EXEC which are deterministically slow. > Pawel, > _______________________________________________ > 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 alexg at etherstack.com Wed Aug 12 00:26:55 2009 From: alexg at etherstack.com (Alex Green) Date: Wed, 12 Aug 2009 17:26:55 +1000 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <20090811141440.106D396C009@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> <20090811141440.106D396C009@etherstack.com> Message-ID: <4A826EBF.20207@etherstack.com> Ok, I think I have gotten to the bottom of it: 1.: Using zap_channel_dequeue_dtmf() correctly identifies digits for both in-band PRI and analog. 2.: Setting an event callback and waiting for ZAP_EVENT_DTMF works for in-band digits on a PRI channel. 3.: Potential bug: As suggested by Moises, adding extra logging in libteletone_detect.c shows that digits were detected, but the ZAP_EVENT_DTMF events were *not* sent back for an analog channel (I am using an Openvox A800p, but it looks to be the same for all analog cards). 4.: Potential bug: DTMF digit detection *eats some audio* each time a digit is detected. If the device is connected directly to free switch this is not a problem. If a device makes a call through freeswitch (to let's say phone banking) a digit that is held down may be detected as 3 digits (sometimes more) due to the consumed audio. This can be heard on the other side of freeswitch as a start-stop at the start of the DTMF. Any thoughts? Cheers -alex Moises Silva wrote: > On Tue, Aug 11, 2009 at 1:17 AM, Alex Green > wrote: > > Thanks Moises, do I need to do a flush_dtmf() or set values for DTMF > on/off period, or anything else? > > > I think default on/off times are good enough, but I suppose you may have > to play with it. As of flushing, sounds like a good idea to do when the > call starts, but that depends on your applications, if you are certain > of consuming all DTMF always, then is not needed. > > Are you using hw dtmf? are you using Sangoma boards? > > I noticed that openzap behavior will be different from an API point of > view when using hardware DTMF than when using software DTMF, since the > callbacks (for span and channel) are currently not available when using > hardware DTMF. That sounds like a bug to me, the behavior should be > available regardless of the backend system used for DTMF. I will fix > that asap. > > > zap_channel_dequeue_dtmf() correctly identifies the digit, but if the > digit is held down it is detected more than once. I am using a frame > size of 160 for the read. > > > More than once like in twice? 3 times, 4 times? ad infinitum if the > digit is hold? I'd need to debug to try to figure out where that comes > from. If you are using hardware dtmf in wanpipe boards you can go to > src/ozmod/ozmod_wanpipe/ozmod_wanpipe.c and search for > WP_TDMAPI_EVENT_DTMF, then uncomment the log line there to see if the > digit is detected twice by the drivers. > > If using software DTMF, some logging in zap_channel_read when calling > teletone_dtmf_detect() can be added to check if its detecting it twice. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 dome at tel.co.th Thu Aug 13 08:26:50 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 13 Aug 2009 22:26:50 +0700 Subject: [Freeswitch-dev] How to contribute ? Message-ID: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> Dear all, I want to contribute mod_say_th (Say for Thai language). how to send patch ? Dome C. From william.suffill at gmail.com Thu Aug 13 08:36:26 2009 From: william.suffill at gmail.com (William Suffill) Date: Thu, 13 Aug 2009 11:36:26 -0400 Subject: [Freeswitch-dev] How to contribute ? In-Reply-To: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> References: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> Message-ID: <6b65470d0908130836j668ad86cg57ce89800f4956a3@mail.gmail.com> Probably best to open a JIRA and attach it. From mike at jerris.com Thu Aug 13 08:41:11 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 13 Aug 2009 11:41:11 -0400 Subject: [Freeswitch-dev] How to contribute ? In-Reply-To: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> References: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> Message-ID: <4CAA305A-AC17-4D99-AA78-BB72FB6FE560@jerris.com> http://jira.freeswitch.org On Aug 13, 2009, at 11:26 AM, Dome Charoenyost wrote: > Dear all, > > I want to contribute mod_say_th (Say for Thai language). how to > send patch ? > > Dome C. From janvb at live.com Thu Aug 13 09:09:52 2009 From: janvb at live.com (Jan Berger) Date: Thu, 13 Aug 2009 18:09:52 +0200 Subject: [Freeswitch-dev] Signalling Status on OpenZap In-Reply-To: <4CAA305A-AC17-4D99-AA78-BB72FB6FE560@jerris.com> References: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> <4CAA305A-AC17-4D99-AA78-BB72FB6FE560@jerris.com> Message-ID: hi, Where do I check out OpenZap? (Yes I know I should remember it) Can anyone give me a status on the ISDN stack - where are we? Jan _________________________________________________________________ Drag n? drop?Get easy photo sharing with Windows Live? Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/b381990f/attachment.html From msc at freeswitch.org Thu Aug 13 09:13:48 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 13 Aug 2009 09:13:48 -0700 Subject: [Freeswitch-dev] Signalling Status on OpenZap In-Reply-To: References: <8ccbff060908130826h7e30f59bn55ad91351cab1bda@mail.gmail.com> <4CAA305A-AC17-4D99-AA78-BB72FB6FE560@jerris.com> Message-ID: <87f2f3b90908130913t17afe7fcgc6c6e2eb58928520@mail.gmail.com> OZ is found in the libs folder of the regular FS SVN checkout. ISDN is still in flux but some of the volunteers have been experimenting. The ozmod_libpri thing is working pretty well for those using it. No ETA on the "pure OZ" PRI stack... -MC On Thu, Aug 13, 2009 at 9:09 AM, Jan Berger wrote: > hi, > > Where do I check out OpenZap? (Yes I know I should remember it) > > Can anyone give me a status on the ISDN stack - where are we? > > Jan > > ------------------------------ > What can you do with the new Windows Live? Find out > > _______________________________________________ > 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/20090813/bd42c158/attachment.html From moises.silva at gmail.com Thu Aug 13 09:24:20 2009 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 13 Aug 2009 12:24:20 -0400 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <4A826EBF.20207@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> <20090811141440.106D396C009@etherstack.com> <4A826EBF.20207@etherstack.com> Message-ID: On Wed, Aug 12, 2009 at 3:26 AM, Alex Green wrote: > 2.: > Setting an event callback and waiting for ZAP_EVENT_DTMF works for > in-band digits on a PRI channel. > Are you just pointing out that it works for PRI or implying that does not work for analog? for hardware DTMF I am sure this will NOT work, either for PRI or Analog. > 3.: Potential bug: > As suggested by Moises, adding extra logging in libteletone_detect.c > shows that digits were detected, but the ZAP_EVENT_DTMF events were > *not* sent back for an analog channel (I am using an Openvox A800p, but > it looks to be the same for all analog cards). > you mean libteletone detects it but ZAP_EVENT_DTMF is not launched? the next test should be in zap_channel_read, which uses teletone_dtmf_detect and teletone_dtmf_get to retrieve DTMF and queue the DTMF, there you will be able to find out what is going on. 4.: Potential bug: > DTMF digit detection *eats some audio* each time a digit is detected. If > the device is connected directly to free switch this is not a problem. > If a device makes a call through freeswitch (to let's say phone banking) > a digit that is held down may be detected as 3 digits (sometimes more) > due to the consumed audio. This can be heard on the other side of > freeswitch as a start-stop at the start of the DTMF. > Indeed ZAP_CHANNEL_SUPRESS_DTMF causes zap_channel_read to memset the read data each time a digit hit is found. The problem is that there is no DTMF_START and DTMF_END events and the DTMF length is lost, we just have a plain DTMF event for the start. Not sure about how to use libteletone to detect the DTMF end though. Anthony, Mike? care to comment about this one? -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/2037753e/attachment.html From kevin.snow at ooma.com Thu Aug 13 10:54:04 2009 From: kevin.snow at ooma.com (Kevin Snow) Date: Thu, 13 Aug 2009 10:54:04 -0700 Subject: [Freeswitch-dev] Answering a call... Message-ID: Guys, In my module (written in C), I have a call answering edge case that I think there must be an easier way than what I?m doing. Or, more correctly, a right way to do this and I?m currently not doing that. I have an active/established call. An incoming SIP call arrives and I send it off to switch_ivr_originate to ring my endpoints. Based on action outside of Freeswitch, I want to stop the ringing and have an endpoint from the established call answer this ringing call. I have a custom FS event generated to indicate this condition. In this event handler I want to take a channel from the active call and answer this incoming call. Here?s what I?m currently doing. In my event handler I?m taking the endpoint of the active call and sending to a specific dialplan This dialplan has it call my C application. At this point the application is the session of the endpoint and it can get the session that?s currently ringing. It gets the session of the ringing channel It uses switch_channel_answer to answer the ringing channel. Then it uses switch_ivr_uuid_bridge to bridge these two endpoints. Seems simple enough but it?s not working. The problems I have are: Doing this seems to make the switch_ivr_originate that was ringing the endpoints fail with ORIGINATOR_CANCEL. I special case this but it leads me to believe I?m going about this the wrong way. I?m not trying to cancel it, I want to answer it. I?m not breaking the bridge of the original established call. That too seems wrong. I think I should be doing something there but it?s not clear what. Is there a better way to handle this? Thanks Kevin Snow -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/f036c708/attachment.html From janvb at live.com Thu Aug 13 12:27:15 2009 From: janvb at live.com (Jan Berger) Date: Thu, 13 Aug 2009 21:27:15 +0200 Subject: [Freeswitch-dev] Skype calls In-Reply-To: References: Message-ID: hi, Does anyone know how many skype calls you can handle on a single PC? I understand that there are a low limit ? Jan _________________________________________________________________ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/7f087669/attachment.html From msc at freeswitch.org Thu Aug 13 12:48:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 13 Aug 2009 14:48:22 -0500 Subject: [Freeswitch-dev] Skype calls In-Reply-To: References:

Message-ID: <87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> On Thu, Aug 13, 2009 at 2:27 PM, Jan Berger wrote: > hi, > > Does anyone know how many skype calls you can handle on a single PC? > > I understand that there are a low limit ? > I believe that Giovanni has done several dozen calls concurrently on a Linux machine. I don't know that anyone has done stress testing to the breaking point. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/bd8668ac/attachment-0001.html From gmaruzz at celliax.org Thu Aug 13 12:50:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 13 Aug 2009 21:50:21 +0200 Subject: [Freeswitch-dev] Skype calls In-Reply-To: References:

Message-ID: <7b197bef0908131250g6fa64775g370be249301c7d6e@mail.gmail.com> Hi Jan, it all boil down to your hardware, cpu and ram. I develop mod_skypiax with 20 concurrent calls on a q6600 with 3GB ram, and I tested 30 concurrent calls without problems. Also the mix OS/Kernel/alsa_version/alsa_driver/ can make a difference. Please, have a look at the wiki page: http://wiki.freeswitch.org/wiki/Skypiax Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Thu, Aug 13, 2009 at 9:27 PM, Jan Berger wrote: > hi, > > Does anyone know how many skype calls you can handle on a single PC? > > I understand that there are a low limit ? > > Jan > > ________________________________ > See all the ways you can stay connected to friends and family > _______________________________________________ > 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 gmaruzz at celliax.org Thu Aug 13 13:01:39 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 13 Aug 2009 22:01:39 +0200 Subject: [Freeswitch-dev] Skype calls In-Reply-To: <87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> References:

<87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> Message-ID: <7b197bef0908131301r5bc4f8a3uea0396c9c1a80367@mail.gmail.com> Hi Michael, Actually I'm working on a kind of stress testing since I come back from Cluecon. First I will commit a big patch to enhance robustness (I'm using sipp to tear up and down 10 calls/sec for 5 secs briding on 20 concurrent skypiax interfaces calling a PSTN number via Skypeout), then I will probably test how many max concurrent calls you can have on my machine. Heh... -giovanni On Thu, Aug 13, 2009 at 9:48 PM, Michael Collins wrote: > > > On Thu, Aug 13, 2009 at 2:27 PM, Jan Berger wrote: >> >> hi, >> >> Does anyone know how many skype calls you can handle on a single PC? >> >> I understand that there are a low limit ? > > I believe that Giovanni has done several dozen calls concurrently on a Linux > machine. I don't know that anyone has done stress testing to the breaking > point. > -MC > > > _______________________________________________ > 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 janvb at live.com Thu Aug 13 13:56:42 2009 From: janvb at live.com (Jan Berger) Date: Thu, 13 Aug 2009 22:56:42 +0200 Subject: [Freeswitch-dev] Skype calls In-Reply-To: <7b197bef0908131301r5bc4f8a3uea0396c9c1a80367@mail.gmail.com> References:

<87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> <7b197bef0908131301r5bc4f8a3uea0396c9c1a80367@mail.gmail.com> Message-ID: hi, 20 calls is very little - any special reason why the limit is so low? Is this CPU or RAM? ------------ Anyway - 20 calls is pretty ok for the Enterprice marked that mainly exist between 1-90 ports + I guess we can look at combining PC's to get more. Another option is to look at interfacing to the lower layer skype protocol - if we can hack it.... just an idea. Jan > From: gmaruzz at celliax.org > Date: Thu, 13 Aug 2009 22:01:39 +0200 > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Skype calls > > Hi Michael, > > Actually I'm working on a kind of stress testing since I come back from Cluecon. > > First I will commit a big patch to enhance robustness (I'm using sipp > to tear up and down 10 calls/sec for 5 secs briding on 20 concurrent > skypiax interfaces calling a PSTN number via Skypeout), then I will > probably test how many max concurrent calls you can have on my > machine. > > Heh... > > -giovanni > > > On Thu, Aug 13, 2009 at 9:48 PM, Michael Collins wrote: > > > > > > On Thu, Aug 13, 2009 at 2:27 PM, Jan Berger wrote: > >> > >> hi, > >> > >> Does anyone know how many skype calls you can handle on a single PC? > >> > >> I understand that there are a low limit ? > > > > I believe that Giovanni has done several dozen calls concurrently on a Linux > > machine. I don't know that anyone has done stress testing to the breaking > > point. > > -MC > > > > > > _______________________________________________ > > 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 _________________________________________________________________ Drag n? drop?Get easy photo sharing with Windows Live? Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/d054435c/attachment.html From mrene_lists at avgs.ca Thu Aug 13 16:41:23 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 13 Aug 2009 19:41:23 -0400 Subject: [Freeswitch-dev] Answering a call... In-Reply-To: References: Message-ID: <2CAE6758-6514-42AA-A5BE-F0387E929C32@avgs.ca> Kevin, From first sight it looks like you aren't bridging the right uuid. Hop on #freeswitch-dev and I can guide you a bit more through the core API. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 13-Aug-09, at 1:54 PM, Kevin Snow wrote: > > Guys, > > In my module (written in C), I have a call answering edge case that > I think there must be an easier way than what I?m doing. Or, more > correctly, a right way to do this and I?m currently not doing that. > > I have an active/established call. An incoming SIP call arrives and > I send it off to switch_ivr_originate to ring my endpoints. Based on > action outside of Freeswitch, I want to stop the ringing and have an > endpoint from the established call answer this ringing call. I have > a custom FS event generated to indicate this condition. In this > event handler I want to take a channel from the active call and > answer this incoming call. > > Here?s what I?m currently doing. > > In my event handler I?m taking the endpoint of the active call and > sending to a specific dialplan > This dialplan has it call my C application. > At this point the application is the session of the endpoint and it > can get the session that?s currently ringing. > It gets the session of the ringing channel > It uses switch_channel_answer to answer the ringing channel. > Then it uses switch_ivr_uuid_bridge to bridge these two endpoints. > > Seems simple enough but it?s not working. > > > The problems I have are: > > Doing this seems to make the switch_ivr_originate that was ringing > the endpoints fail with ORIGINATOR_CANCEL. I special case this but > it leads me to believe I?m going about this the wrong way. I?m not > trying to cancel it, I want to answer it. > > I?m not breaking the bridge of the original established call. That > too seems wrong. I think I should be doing something there but it?s > not clear what. > > > Is there a better way to handle this? > > > Thanks > > Kevin Snow > > > > _______________________________________________ > 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/20090813/bbf4eb10/attachment.html From msc at freeswitch.org Thu Aug 13 17:43:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 13 Aug 2009 19:43:25 -0500 Subject: [Freeswitch-dev] Skype calls In-Reply-To: References:

<87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> <7b197bef0908131301r5bc4f8a3uea0396c9c1a80367@mail.gmail.com> Message-ID: <87f2f3b90908131743q7ca27ae9y782ac5398c7aea78@mail.gmail.com> On Thu, Aug 13, 2009 at 3:56 PM, Jan Berger wrote: > hi, > > 20 calls is very little - any special reason why the limit is so low? Is > this CPU or RAM? > I suppose it all goes back to the fact that each Skype call requires its own instance of the Skype client. Unless the Skype client is REALLY efficient then that can scale only so high. The permanent solution will be when Skype breaks down and let's us have a lib that allows the devs to create a proper channel driver and codec handler. Until then you'll need to throw CPU at the problem... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090813/61370241/attachment.html From gmaruzz at celliax.org Thu Aug 13 22:09:24 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 14 Aug 2009 07:09:24 +0200 Subject: [Freeswitch-dev] Skype calls In-Reply-To: <87f2f3b90908131743q7ca27ae9y782ac5398c7aea78@mail.gmail.com> References:

<87f2f3b90908131248x40aeb9a6hafa8af5f6de3b7be@mail.gmail.com> <7b197bef0908131301r5bc4f8a3uea0396c9c1a80367@mail.gmail.com> <87f2f3b90908131743q7ca27ae9y782ac5398c7aea78@mail.gmail.com> <7b197bef0908132209t4883dd2blbaa30491462d9105@mail.gmail.com> <7b197bef0908132308rfb81198we0aef2c74ba8329f@mail.gmail.com> Message-ID: hi, Its many ways around this... But first - You have to take into account that Skype might not be interested in pabx type connections (for now) as theire API only provide a per line interface. Enabling pabx - multiple connections also means someone else can clone skype using skypes own business... Jan > From: gmaruzz at celliax.org > Date: Fri, 14 Aug 2009 08:08:47 +0200 > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Skype calls > > Muhammad, > > Yes, I can confirm the behavior of the echo123 service. Only one call > from each username. > > For my massive tests, I use some tollfree PSTN number via skypeout > (you can do it also without buying credits). > > -giovanni > > On Fri, Aug 14, 2009 at 8:04 AM, Muhammad > Shahzad wrote: > > I have bring up to 60 Skypiax channels, using 20 skype accounts, i.e. each > > skype account logged in three times. But only 20 of them were able to make > > call to skype echo test service, others got NO ANSWER. I am not sure of why > > this happened but may be skype echo service allows only one call per skype > > account regardless of how may instances are using this one account. Has > > anyone tried this with different results? > > > > Thank you. > > > > > > On Fri, Aug 14, 2009 at 11:09 AM, Giovanni Maruzzelli > > wrote: > >> > >> On Fri, Aug 14, 2009 at 2:43 AM, Michael Collins > >> wrote: > >> > I suppose it all goes back to the fact that each Skype call requires its > >> > own > >> > instance of the Skype client. Unless the Skype client is REALLY > >> > efficient > >> > then that can scale only so high. The permanent solution will be when > >> > Skype > >> > >> The Skype client is a pig, the endpoint (channel driver) use one Skype > >> client per call because it is the ONLY legal way to connect to the > >> Skype network (eg: is illegal to reverse engineer, etc), a lot of > >> effort has been put in understanding how to lower the CPU usage by the > >> Skype client. > >> > >> Have a look at the wiki page http://wiki.freeswitch.org/wiki/Skypiax > >> > >> As soon as people will put pressure on Skype/Ebay/Whoever owns the > >> underlying technology so they open and document the protocol, we will > >> implement an endpoint that does not depend from the client. > >> > >> Btw, will also be a step forward if they will give API access to the > >> library underlying the Skype client, but maybe that one is makeing the > >> Skype client so a pig :-). > >> > >> -giovanni > >> > >> _______________________________________________ > >> 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 > > > > > > > > -- > > Muhammad Shahzad > > ----------------------------------- > > CISCO Rich Media Communication Specialist (CRMCS) > > CISCO Certified Network Associate (CCNA) > > Cell: +92 334 422 40 88 > > MSN: shari_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > > > _______________________________________________ > > 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 _________________________________________________________________ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090814/5d492624/attachment.html From gmaruzz at celliax.org Fri Aug 14 10:43:27 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 14 Aug 2009 19:43:27 +0200 Subject: [Freeswitch-dev] Skypiax, Skype endpoint and trunk, robustness patch Message-ID: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> Hi FreeSWITCHers, all the users of mod_skypiax are kindly requested to test the svn trunk 14519. It contains a lot of changes meant to add stability and robustness, toward a production environment. Let me know how your feelings, and please add to the Jira any possible bug/issue/etc. Thanks to you all, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at celliax.org Fri Aug 14 14:02:06 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 14 Aug 2009 23:02:06 +0200 Subject: [Freeswitch-dev] Skypiax, Skype endpoint and trunk, robustness patch In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> Message-ID: <7b197bef0908141402k15d4144bt87a287e07c51b5c6@mail.gmail.com> svn 14521: skypiax: compiles on windoz, not yet tested (on windoz) On Fri, Aug 14, 2009 at 7:43 PM, Giovanni Maruzzelli wrote: > Hi FreeSWITCHers, > > all the users of mod_skypiax are kindly requested to test the svn trunk 14519. > > It contains a lot of changes meant to add stability and robustness, > toward a production environment. > > Let me know how your feelings, and please add to the Jira any possible > bug/issue/etc. > > Thanks to you all, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > > Cell : +39-347-2665618 > From gilles.depatie at mapleworks.com Mon Aug 17 06:49:01 2009 From: gilles.depatie at mapleworks.com (Gilles Depatie) Date: Mon, 17 Aug 2009 09:49:01 -0400 Subject: [Freeswitch-dev] RFC 2198 (RTP Payload for Redundant Audio Data) Message-ID: <26A787B362F14E02A8FD1D24C77C0CFF@mapleworks.com> Hi Has anyone considered or attempted to implement this piece of functionality? As part of contract work for a customer using FreeSwitch, we have been asked to enhance FS to support this RFC. Any help or comments would be appreciated. Thanks Gilles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090817/e226db4e/attachment-0001.html From santosh_tripathi at datamatics.com Wed Aug 19 03:55:11 2009 From: santosh_tripathi at datamatics.com (Santosh) Date: Wed, 19 Aug 2009 16:25:11 +0530 Subject: [Freeswitch-dev] callerid Message-ID: <23e501ca20bb$8669bc10$1805a8c0@tes.datamatics.com> Hi, I am new to freeswitch and i need to fetch the callerid information.But my freeswitch is not showing me the callerid not even in the log.For the client I am using Linphone. How am I suppose to activate it.Your help would be appreciated. Regards, Santosh Disclaimer: The information contained in this e-mail and attachments if any are privileged and confidential and are intended for the individual(s) or entity(ies) named in this e-mail. If the reader or recipient is not the intended recipient, or employee or agent responsible for delivering to the intended recipient, you are hereby notified that dissemination, distribution or copying of this communication or attachments thereof is strictly prohibited. IF YOU RECEIVE this communication in error, please immediately notify the sender and return the original message. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090819/73153a0f/attachment.html From msc at freeswitch.org Wed Aug 19 09:38:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 19 Aug 2009 09:38:41 -0700 Subject: [Freeswitch-dev] callerid In-Reply-To: <23e501ca20bb$8669bc10$1805a8c0@tes.datamatics.com> References: <23e501ca20bb$8669bc10$1805a8c0@tes.datamatics.com> Message-ID: <87f2f3b90908190938x306538dbu1695179250ac7c9d@mail.gmail.com> On Wed, Aug 19, 2009 at 3:55 AM, Santosh wrote: > Hi, > I am new to freeswitch and i need to fetch the callerid information.But my > freeswitch is not showing me the callerid not even in the log.For the client > I am using Linphone. > How am I suppose to activate it.Your help would be appreciated. > Can you elaborate a little bit? What kind of call? Inbound/outbound? Between to registered endpoints? Did you make any modifications to the default configuration? Thanks, the details will help. I also recommend that you go to the FS pastebin and paste the debug output from a sample call. ( http://www.pastebin.com - and be sure to read the box very carefully when it asks you for a username and password. ;) ) -MC > > Regards, > Santosh > > > > > > Disclaimer: The information contained in this e-mail and attachments if any > are privileged and confidential and are intended for the individual(s) or > entity(ies) named in this e-mail. If the reader or recipient is not the > intended recipient, or employee or agent responsible for delivering to the > intended recipient, you are hereby notified that dissemination, distribution > or copying of this communication or attachments thereof is strictly > prohibited. IF YOU RECEIVE this communication in error, please immediately > notify the sender and return the original message. > > > > _______________________________________________ > 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/20090819/7070eab1/attachment.html From mrene_lists at avgs.ca Wed Aug 19 09:45:21 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 19 Aug 2009 12:45:21 -0400 Subject: [Freeswitch-dev] callerid In-Reply-To: <87f2f3b90908190938x306538dbu1695179250ac7c9d@mail.gmail.com> References: <23e501ca20bb$8669bc10$1805a8c0@tes.datamatics.com> <87f2f3b90908190938x306538dbu1695179250ac7c9d@mail.gmail.com> Message-ID: <91AF18F0-1074-4020-AEBC-FA2BAC1C1CE0@avgs.ca> You may also have more luck on freeswitch-users on general usage questions. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 19-Aug-09, at 12:38 PM, Michael Collins wrote: > On Wed, Aug 19, 2009 at 3:55 AM, Santosh > wrote: > Hi, > I am new to freeswitch and i need to fetch the callerid > information.But my freeswitch is not showing me the callerid not > even in the log.For the client I am using Linphone. > How am I suppose to activate it.Your help would be appreciated. > > Can you elaborate a little bit? What kind of call? Inbound/outbound? > Between to registered endpoints? Did you make any modifications to > the default configuration? > > Thanks, the details will help. I also recommend that you go to the > FS pastebin and paste the debug output from a sample call. (http://www.pastebin.com > - and be sure to read the box very carefully when it asks you for a > username and password. ;) ) > > -MC > > > Regards, > Santosh > > > > > > Disclaimer: The information contained in this e-mail and attachments > if any are privileged and confidential and are intended for the > individual(s) or entity(ies) named in this e-mail. If the reader or > recipient is not the intended recipient, or employee or agent > responsible for delivering to the intended recipient, you are hereby > notified that dissemination, distribution or copying of this > communication or attachments thereof is strictly prohibited. IF YOU > RECEIVE this communication in error, please immediately notify the > sender and return the original message. > > > > _______________________________________________ > 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/20090819/ad547d9a/attachment.html From hai.lei at mapleworks.com Wed Aug 19 14:00:16 2009 From: hai.lei at mapleworks.com (hai lei) Date: Wed, 19 Aug 2009 17:00:16 -0400 Subject: [Freeswitch-dev] Event Socket issue. Message-ID: <7BE4D715655E4AFEBD2D6644443E8D34@mapleworks.com> Hello, I am using Freeswitch to do some development work and encounter a problem with Event Socket. The configuration is: Freeswitch works as PBX. I also configure three softphones A, B and C. What I want is: 1. A call B. 2. B reject the call. 3. Freeswitch start conference involve A, B and C. What I did was: 1. Add perl script to the dialplan, so the perl script will be called after B reject the call from A. Worked fine. 2. In the script, I connected to localhost:8021 so the Freeswitch work in Inbound mode. Worked fine. 3. Then run $connection->execute to run "conference_set_auto_outcall" and "conference" command. Not worked. So I check the "parse_command" function in file "mod_event_socket.c", the following is what happened: Since I give the uuid to "$connection->execute" command, so FS doesn't care if it is Async mode or not. Actually FS just call function "switch_core_session_queue_private_event" to queue the event. Here is the problem, since B already reject the call, then nobody will call the function "switch_ivr_parse_event" to execute the command. I did try to add code to "parse_command" function to treat the Async and Sync mode differently when the uuid is provided. The conference was started successfully in Sync mode but since the "conference" application is blocked, my perl script was not able to move on since it was waiting for the "conference" to stop. If I choose "$connection->executeAsync", the commands were not run because of the reason I mentioned above. What is the proper solution for this? Thanks Hai Lei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090819/acffa159/attachment.html From alexg at etherstack.com Wed Aug 19 22:58:49 2009 From: alexg at etherstack.com (Alex Green) Date: Thu, 20 Aug 2009 15:58:49 +1000 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <20090813163404.42E34774004@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> <20090811141440.106D396C009@etherstack.com> <4A826EBF.20207@etherstack.com> <20090813163404.42E34774004@etherstack.com> Message-ID: <4A8CE619.50502@etherstack.com> Any thought on the email below? I would imagine that passing an analog call through freeswitch is not too uncommon a use-case... > 4.: Potential bug: > DTMF digit detection *eats some audio* each time a digit is detected. If > the device is connected directly to free switch this is not a problem. > If a device makes a call through freeswitch (to let's say phone banking) > a digit that is held down may be detected as 3 digits (sometimes more) > due to the consumed audio. This can be heard on the other side of > freeswitch as a start-stop at the start of the DTMF. > > > Indeed ZAP_CHANNEL_SUPRESS_DTMF causes zap_channel_read to memset the > read data each time a digit hit is found. The problem is that there is > no DTMF_START and DTMF_END events and the DTMF length is lost, we just > have a plain DTMF event for the start. Not sure about how to use > libteletone to detect the DTMF end though. > DTMF suppression is mangling the digits. It took me a while to get to the bottom of this, but it turns out to be very easy to reproduce. Take Freeswitch with an FXS port and a handset. Dial 1000 on the handset and answer the call on a SIP client. Now hold down a digit. You will notice that the digit has a wobble at the start. This is the symptom. Commenting out the DTMF suppression code, changes the behaviour but does not fix the problem. This is what baffled me for a while. I wrote some trivial code that bridges two analog channels by doing a zap_channel_read/write() from one to the other. Commenting out the memset() on line 2042 of zap_io.c produced perfect digits while uncommenting it produced the wobble. The obvious conclusion is that the dodgy digit suppression is being attempted somewhere else (like the SIP side?) and thus the wobble is being introduced at more than one place. This also explains why I (non scientifically) noticed that the more times the call passed through Freeswitch the more incorrectly recognised digits were reported at the final point. I imagine that a solution might be to just turn off DTMF digit suppression by default until it is fixed. Cheers, -alex > Anthony, Mike? care to comment about this one? > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com From juanbackson at gmail.com Thu Aug 20 05:50:04 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 20 Aug 2009 20:50:04 +0800 Subject: [Freeswitch-dev] Unable to access directory variable from channel Message-ID: <27c25bc40908200550h27bf96dfrf0ba1833ca0619b9@mail.gmail.com> Hi, I set up a variable called "account-id" in the user directory xml file. >From the info app, I can't find the variable for variable_account-id. Also, in my C module, I tried to obtain the value with switch_channel_get_variable(channel,"account-id") and it is not returning null. Does anyone know how I can access the user variable from within my mod? Thanks, JB -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090820/a01cc5c3/attachment.html From mrene_lists at avgs.ca Thu Aug 20 06:22:54 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 20 Aug 2009 09:22:54 -0400 Subject: [Freeswitch-dev] Unable to access directory variable from channel In-Reply-To: <27c25bc40908200550h27bf96dfrf0ba1833ca0619b9@mail.gmail.com> References: <27c25bc40908200550h27bf96dfrf0ba1833ca0619b9@mail.gmail.com> Message-ID: <5FD86C1A-EA80-40EB-8E93-4FB112041D6F@avgs.ca> Hi, switch_channel_get_variable() is the right way. Its possible that the user isn't recognized, pastebin some debug logs. (console loglevel debug) PS: you can hop on #freeswitch-dev and we can chat Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 20-Aug-09, at 8:50 AM, Juan Backson wrote: > Hi, > > I set up a variable called "account-id" in the user directory xml > file. > > From the info app, I can't find the variable for variable_account- > id. Also, in my C module, I tried to obtain the value with > switch_channel_get_variable(channel,"account-id") and it is not > returning null. > > Does anyone know how I can access the user variable from within my > mod? > > Thanks, > JB > _______________________________________________ > 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 moises.silva at gmail.com Thu Aug 20 10:03:30 2009 From: moises.silva at gmail.com (Moises Silva) Date: Thu, 20 Aug 2009 13:03:30 -0400 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <4A8CE619.50502@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> <20090811141440.106D396C009@etherstack.com> <4A826EBF.20207@etherstack.com> <20090813163404.42E34774004@etherstack.com> <4A8CE619.50502@etherstack.com> Message-ID: I discussed with Mike Jerris some days ago, and as I pointed out we need to be able to tell the duration of the DTMF. At this point libteletone does not help with that (AFAIK), so we need to investigate and fix it. Just haven't found the time to do it. Help is welcomed :-) On Thu, Aug 20, 2009 at 1:58 AM, Alex Green wrote: > Any thought on the email below? I would imagine that passing an analog > call through freeswitch is not too uncommon a use-case... > > > > 4.: Potential bug: > > DTMF digit detection *eats some audio* each time a digit is detected. > If > > the device is connected directly to free switch this is not a > problem. > > If a device makes a call through freeswitch (to let's say phone > banking) > > a digit that is held down may be detected as 3 digits (sometimes > more) > > due to the consumed audio. This can be heard on the other side of > > freeswitch as a start-stop at the start of the DTMF. > > > > > > Indeed ZAP_CHANNEL_SUPRESS_DTMF causes zap_channel_read to memset the > > read data each time a digit hit is found. The problem is that there is > > no DTMF_START and DTMF_END events and the DTMF length is lost, we just > > have a plain DTMF event for the start. Not sure about how to use > > libteletone to detect the DTMF end though. > > > DTMF suppression is mangling the digits. > It took me a while to get to the bottom of this, but it turns out to be > very easy to reproduce. Take Freeswitch with an FXS port and a handset. > Dial 1000 on the handset and answer the call on a SIP client. Now hold > down a digit. You will notice that the digit has a wobble at the start. > This is the symptom. > > Commenting out the DTMF suppression code, changes the behaviour but does > not fix the problem. This is what baffled me for a while. I wrote some > trivial code that bridges two analog channels by doing a > zap_channel_read/write() from one to the other. Commenting out the > memset() on line 2042 of zap_io.c produced perfect digits while > uncommenting it produced the wobble. The obvious conclusion is that the > dodgy digit suppression is being attempted somewhere else (like the SIP > side?) and thus the wobble is being introduced at more than one place. > This also explains why I (non scientifically) noticed that the more > times the call passed through Freeswitch the more incorrectly recognised > digits were reported at the final point. > > I imagine that a solution might be to just turn off DTMF digit > suppression by default until it is fixed. > > Cheers, -alex > > > > Anthony, Mike? care to comment about this one? > > > > -- > > Moises Silva > > Software Developer > > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > > 9T3 Canada > > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > _______________________________________________ > 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 > -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090820/b9447dcb/attachment.html From gilles.depatie at mapleworks.com Thu Aug 20 14:54:34 2009 From: gilles.depatie at mapleworks.com (Gilles Depatie) Date: Thu, 20 Aug 2009 17:54:34 -0400 Subject: [Freeswitch-dev] Debugging Question Message-ID: <423F973F6C4246B283208E94C1EA4C62@mapleworks.com> Hi I'm new to Freeswitch environment, but I have attempted to build an executable where I could set breakpoints etc. I have followed the instructions on the wiki where I define the two environment vars CFLAGS and MOD_CFLAGS as "-g -ggdb" I then run configure which sets the gcc compile option parameters across all makefiles. I notice that during the compilation, those option are invoked for all source file that have produce output. I've also confirmed that the object files for a sample set of files are slightly larger when the compiler output has the options set than compiling without those two options. So I assume that debug symbols were produced when gcc is invoked with the debug option. What I see is that after I have run make install, the freeswitch executable is not any larger when debug symbols are requested than when it is not. In fact the opposite is true. Size with debug = "114804", WITHOUT debug = "114908" When I attempt to list files from the GDB command line, I get the following: "No source file named xxx.c" The only file I can list is "switch.c" , which is the one that has the main program. Here is my environment: VmWare Server 2.0 Red Hat 9 Linux, Kernel 2.6 FreeSwitch-1.0.4 gcc version 3.2.2 20030222 GNU gdb Red Hat Linux (5.3post-0.20021129.18rh) Any help would be appreciated Thanks Gilles Depatie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090820/a4d8dcd1/attachment-0001.html From mrene_lists at avgs.ca Thu Aug 20 15:16:04 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 20 Aug 2009 18:16:04 -0400 Subject: [Freeswitch-dev] Debugging Question In-Reply-To: <423F973F6C4246B283208E94C1EA4C62@mapleworks.com> References: <423F973F6C4246B283208E94C1EA4C62@mapleworks.com> Message-ID: Hi, FreeSWITCH builds with debug symbols with the default configuration, but the source file has to be in the same path as when it was compiled. If you wouldnt have debug symbols in, it wouldn't know the file & location of the function you are trying to see. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 20-Aug-09, at 5:54 PM, Gilles Depatie wrote: > Hi > > I?m new to Freeswitch environment, but I have attempted to build an > executable where I could set breakpoints etc. > > I have followed the instructions on the wiki where I define the two > environment vars CFLAGS and MOD_CFLAGS as ?-g ?ggdb? > > I then run configure which sets the gcc compile option parameters > across all makefiles. > > I notice that during the compilation, those option are invoked for > all source file that have produce output. > > I?ve also confirmed that the object files for a sample set of files > are slightly larger when the compiler output has the options set > than compiling without those two options. So I assume that debug > symbols were produced when gcc is invoked with the debug option. > > What I see is that after I have run make install, the freeswitch > executable is not any larger when debug symbols are requested than > when it is not. In fact the opposite is true. Size with debug = > ?114804?, WITHOUT debug = ?114908? > > When I attempt to list files from the GDB command line, I get the > following: ?No source file named xxx.c? > > The only file I can list is ?switch.c? , which is the one that has > the main program. > > > > Here is my environment: > > VmWare Server 2.0 > > Red Hat 9 Linux, Kernel 2.6 > > FreeSwitch-1.0.4 > > gcc version 3.2.2 20030222 > > GNU gdb Red Hat Linux (5.3post-0.20021129.18rh) > > Any help would be appreciated > > Thanks > > Gilles Depatie > _______________________________________________ > 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/20090820/dd5ada14/attachment.html From alexg at etherstack.com Thu Aug 20 19:07:50 2009 From: alexg at etherstack.com (Alex Green) Date: Fri, 21 Aug 2009 12:07:50 +1000 Subject: [Freeswitch-dev] Detecting in-band DTMF Digits in Openzap? In-Reply-To: <20090820171739.0D3A096C003@etherstack.com> References: <4A80D6C0.8050409@etherstack.com> <20090811040854.16C4096C001@etherstack.com> <4A80FF06.5010208@etherstack.com> <20090811141440.106D396C009@etherstack.com> <4A826EBF.20207@etherstack.com> <20090813163404.42E34774004@etherstack.com> <4A8CE619.50502@etherstack.com> <20090820171739.0D3A096C003@etherstack.com> Message-ID: <4A8E0176.9060606@etherstack.com> Thanks Moises, I am keen to help. Implementing this would require architectural changes, which would be better made by people closer to and more experienced with freeswitch. Is there anything wrong with simply disabling DTMF suppression? All scenarios I can think of it is not going to matter if the in-band digits are passed through whole. The dirty hack is to comment out the memset line 2042 of zap_io.c, but something is also going on in the IVR bridge. I know line 261 and onwards in switch_ivr_bridge.c has something to do with it, but I do not understand the IVR stuff enough offer anything useful. As a test (not a fix!) commenting out the while loop on line 261 (in addition to the memset in zap_io.c) fixes the symptoms. Cheers -alex Moises Silva wrote: > I discussed with Mike Jerris some days ago, and as I pointed out we need > to be able to tell the duration of the DTMF. At this point libteletone > does not help with that (AFAIK), so we need to investigate and fix it. > Just haven't found the time to do it. Help is welcomed :-) > > On Thu, Aug 20, 2009 at 1:58 AM, Alex Green > wrote: > > Any thought on the email below? I would imagine that passing an analog > call through freeswitch is not too uncommon a use-case... > > > > 4.: Potential bug: > > DTMF digit detection *eats some audio* each time a digit is > detected. If > > the device is connected directly to free switch this is not a > problem. > > If a device makes a call through freeswitch (to let's say > phone banking) > > a digit that is held down may be detected as 3 digits > (sometimes more) > > due to the consumed audio. This can be heard on the other side of > > freeswitch as a start-stop at the start of the DTMF. > > > > > > Indeed ZAP_CHANNEL_SUPRESS_DTMF causes zap_channel_read to memset the > > read data each time a digit hit is found. The problem is that > there is > > no DTMF_START and DTMF_END events and the DTMF length is lost, we > just > > have a plain DTMF event for the start. Not sure about how to use > > libteletone to detect the DTMF end though. > > > DTMF suppression is mangling the digits. > It took me a while to get to the bottom of this, but it turns out to be > very easy to reproduce. Take Freeswitch with an FXS port and a handset. > Dial 1000 on the handset and answer the call on a SIP client. Now hold > down a digit. You will notice that the digit has a wobble at the start. > This is the symptom. > > Commenting out the DTMF suppression code, changes the behaviour but does > not fix the problem. This is what baffled me for a while. I wrote some > trivial code that bridges two analog channels by doing a > zap_channel_read/write() from one to the other. Commenting out the > memset() on line 2042 of zap_io.c produced perfect digits while > uncommenting it produced the wobble. The obvious conclusion is that the > dodgy digit suppression is being attempted somewhere else (like the SIP > side?) and thus the wobble is being introduced at more than one place. > This also explains why I (non scientifically) noticed that the more > times the call passed through Freeswitch the more incorrectly recognised > digits were reported at the final point. > > I imagine that a solution might be to just turn off DTMF digit > suppression by default until it is fixed. > > Cheers, -alex > > > > Anthony, Mike? care to comment about this one? > > > > -- > > Moises Silva > > Software Developer > > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham > ON L3R > > 9T3 Canada > > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > > _______________________________________________ > 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 > > > > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R > 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 rs at runsolutions.com Fri Aug 21 04:27:10 2009 From: rs at runsolutions.com (Raimund Sacherer) Date: Fri, 21 Aug 2009 13:27:10 +0200 Subject: [Freeswitch-dev] New member Message-ID: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> Hello Freeswitch Developers, I work for an linux solution provider in spain and have lot's of experience in developing for opensource software (freeradius) and adminstration. We have quite a few clients with asterisk solutions and some pretty cool projects in our queue but we have expierenced a lot of problems with asterisk. After reading the freeswitch introduction from Anthony Minessale i at once felt at home with freeswitch as the project seems to be what I wanted from asterisk. As i am a very deep debian enthusiast I want to bring the debian package into debian itself, which means bringing it policy compliant. Which as one of the more problematic moves means: changing all directorys to be debian compliant. i really would love to do this officially as the debian maintainer for the freeswitch project, i already sent the current packet builder an email but I did not receive an answer until know. I hope we can work together and that i will be accepted into this project which has in my oppinion a very bright future in the telecomm business! all the best, -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares From mike at jerris.com Fri Aug 21 05:42:19 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Aug 2009 08:42:19 -0400 Subject: [Freeswitch-dev] New member In-Reply-To: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> Message-ID: On Aug 21, 2009, at 7:27 AM, Raimund Sacherer wrote: > Hello Freeswitch Developers, > > I work for an linux solution provider in spain and have lot's of > experience in developing for opensource software (freeradius) and > adminstration. We have quite a few clients with asterisk solutions and > some pretty cool projects in our queue but we have expierenced a lot > of problems with asterisk. > > After reading the freeswitch introduction from Anthony Minessale i at > once felt at home with freeswitch as the project seems to be what I > wanted from asterisk. > > As i am a very deep debian enthusiast I want to bring the debian > package into debian itself, which means bringing it policy compliant. > Which as one of the more problematic moves means: > > changing all directorys to be debian compliant. We decided so that it is possible to support many different operating systems, we would make all our packages use the same path structure like we do today. To change it would add a burden on us for support and we don't plan on doing that. > i really would love to do this officially as the debian maintainer for > the freeswitch project, i already sent the current packet builder an > email but I did not receive an answer until know. Feel free to toss patches at us via jira.freeswitch.org. We need to start getting everything in place for 1.0.5. > I hope we can work together and that i will be accepted into this > project which has in my oppinion a very bright future in the telecomm > business! Thanks. Also, feel free to catch up with us on irc to discuss any of these issues. From mike at jerris.com Fri Aug 21 06:26:14 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 21 Aug 2009 09:26:14 -0400 Subject: [Freeswitch-dev] New member In-Reply-To: References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> Message-ID: <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> On Aug 21, 2009, at 8:42 AM, Michael Jerris wrote: > > On Aug 21, 2009, at 7:27 AM, Raimund Sacherer wrote: > >> Hello Freeswitch Developers, >> >> I work for an linux solution provider in spain and have lot's of >> experience in developing for opensource software (freeradius) and >> adminstration. We have quite a few clients with asterisk solutions >> and >> some pretty cool projects in our queue but we have expierenced a lot >> of problems with asterisk. >> >> After reading the freeswitch introduction from Anthony Minessale i at >> once felt at home with freeswitch as the project seems to be what I >> wanted from asterisk. >> >> As i am a very deep debian enthusiast I want to bring the debian >> package into debian itself, which means bringing it policy compliant. >> Which as one of the more problematic moves means: >> >> changing all directorys to be debian compliant. > > We decided so that it is possible to support many different > operating systems, we would make all our packages use the same path > structure like we do today. To change it would add a burden on us > for support and we don't plan on doing that. For more information on our approach check out: http://www.pathname.com/fhs/pub/fhs-2.3.html#OPTADDONAPPLICATIONSOFTWAREPACKAGES http://refspecs.freestandards.org/LSB_4.0.0/LSB-Core-generic/LSB-Core-generic.txt 17.1.8. Installable applications > >> i really would love to do this officially as the debian maintainer >> for >> the freeswitch project, i already sent the current packet builder an >> email but I did not receive an answer until know. > > Feel free to toss patches at us via jira.freeswitch.org. We need to > start getting everything in place for 1.0.5. > >> I hope we can work together and that i will be accepted into this >> project which has in my oppinion a very bright future in the telecomm >> business! > > Thanks. Also, feel free to catch up with us on irc to discuss any > of these issues. From rs at runsolutions.com Mon Aug 24 04:16:33 2009 From: rs at runsolutions.com (Raimund Sacherer) Date: Mon, 24 Aug 2009 13:16:33 +0200 Subject: [Freeswitch-dev] New member In-Reply-To: References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> Message-ID: <96515ED9-DE2A-438C-9790-33CAEAF1AE6E@runsolutions.com> Hello, First I contacted the debian Maintainer who tried packaging freeswitch once, but because he does not have the time the package has not been touched in quite a time. Until now I did not get any response so I will wait a little further and then talk with the debian quality assurance team. On Aug 21, 2009, at 2:42 PM, Michael Jerris wrote: >> >> changing all directorys to be debian compliant. > > We decided so that it is possible to support many different operating > systems, we would make all our packages use the same path structure > like we do today. To change it would add a burden on us for support > and we don't plan on doing that. Ok, I was thinking about this about the weekend, I'l have to talk to the debian group as well but I'l have an idea: to be debian compliant AND to not interrupt with the way you provide your package we could do: * Implement all packages debian specific (pathes, etc.) * Create a freeswitch-pathcompatibility package which creates links to /opt so scripts/users which are allready acustomed to your scheme can have it their way, and the debianers out there which love their debian conform system can simply deinstall / not install the pathcompatibility package and everyone should be happy! What do you think of this idea? > >> i really would love to do this officially as the debian maintainer >> for >> the freeswitch project, i already sent the current packet builder an >> email but I did not receive an answer until know. > > Feel free to toss patches at us via jira.freeswitch.org. We need to > start getting everything in place for 1.0.5. Ok, i will doing my first adaptions starting this week. > >> I hope we can work together and that i will be accepted into this >> project which has in my oppinion a very bright future in the telecomm >> business! > > Thanks. Also, feel free to catch up with us on irc to discuss any of > these issues. I will do this, i am Hatrix (registered nick) in IRC, thanks, best -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/ffe8b149/attachment.html From Suneel.Papineni at mettoni.com Mon Aug 24 07:05:23 2009 From: Suneel.Papineni at mettoni.com (Suneel Papineni) Date: Mon, 24 Aug 2009 15:05:23 +0100 Subject: [Freeswitch-dev] Error in call transfer References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> Message-ID: <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> Hi Friends, I just started using Freeswitch for couple of ideas to try and test. I am testing the installation in Windows Vista system. Tried to build from Freeswitch.2008.sln, but failed as proper environment is not there in the system. As I want to test the functionality, 1. I have downloaded Precompiled Binary from "http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe" and installed. Installation went fine and couple of call were tried using X-Lite. 2. Calls are working fine between the extensions. 3. Tried to test to test the functionality of IVR and called 5000 from extension 1001. Respective Welcome message and other files are played. 4. Tried to transfer the call to another extension 1002, it thrown an error and a message is played saying "the extension dialled 1002 is not currently logged in" and call got hang up. Please find attached the logs and screenshot of the error. Please help me in understanding this issue and how to resolve this. Thanks in advance. Thanks & Regards Suny ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN ************************************************************************* -------------- next part -------------- A non-text attachment was scrubbed... Name: Error_with_5000_IVRcalling.jpg Type: image/jpeg Size: 188307 bytes Desc: Error_with_5000_IVRcalling.jpg Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/aa7c4555/attachment-0001.jpg -------------- next part -------------- A non-text attachment was scrubbed... Name: Error_with_5000_IVRcalling.rtf Type: application/rtf Size: 2927 bytes Desc: Error_with_5000_IVRcalling.rtf Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/aa7c4555/attachment-0001.rtf From brian at freeswitch.org Mon Aug 24 08:00:12 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 24 Aug 2009 10:00:12 -0500 Subject: [Freeswitch-dev] Error in call transfer In-Reply-To: <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> Message-ID: You seem to not have the MOH files installed. In addition you seem to have hijacked the previous thread. This happens when you click reply to a message, change the subject and the body and start your message... That will cause the previous thread to be hijacked... Please click new message and input the freeswitch-dev at lists.freeswitch.org as the address then type your message as to prevent any confusion. Sometimes your message might get lost in a thread that someone has already chosen to ignore because it doesn't interest them. Thanks, Brian On Aug 24, 2009, at 9:05 AM, Suneel Papineni wrote: > Hi Friends, > > I just started using Freeswitch for couple of ideas to try and test. > > I am testing the installation in Windows Vista system. Tried to build > from Freeswitch.2008.sln, but failed as proper environment is not > there > in the system. As I want to test the functionality, > > 1. I have downloaded Precompiled Binary from > "http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe" > and > installed. Installation went fine and couple of call were tried using > X-Lite. > 2. Calls are working fine between the extensions. > 3. Tried to test to test the functionality of IVR and called 5000 from > extension 1001. Respective Welcome message and other files are played. > 4. Tried to transfer the call to another extension 1002, it thrown an > error and a message is played saying "the extension dialled 1002 is > not > currently logged in" and call got hang up. > > Please find attached the logs and screenshot of the error. Please help > me in understanding this issue and how to resolve this. Thanks in > advance. > > Thanks & Regards > Suny -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/a9acd982/attachment.html From anthony.minessale at gmail.com Mon Aug 24 09:00:45 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 24 Aug 2009 11:00:45 -0500 Subject: [Freeswitch-dev] New member In-Reply-To: <96515ED9-DE2A-438C-9790-33CAEAF1AE6E@runsolutions.com> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <96515ED9-DE2A-438C-9790-33CAEAF1AE6E@runsolutions.com> Message-ID: <191c3a030908240900s5dd242a2k7572ecd51f6bddfd@mail.gmail.com> You are welcome to the job. The biggest problem we have had is lack of interest in these maintenance positions in the community. On Mon, Aug 24, 2009 at 6:16 AM, Raimund Sacherer wrote: > > Hello, > First I contacted the debian Maintainer who tried packaging freeswitch > once, but because he does not have the time the package has not been touched > in quite a time. Until now I did not get any response so I will wait a > little further and then talk with the debian quality assurance team. > > On Aug 21, 2009, at 2:42 PM, Michael Jerris wrote: > > > changing all directorys to be debian compliant. > > > We decided so that it is possible to support many different operating > systems, we would make all our packages use the same path structure > like we do today. To change it would add a burden on us for support > and we don't plan on doing that. > > > Ok, I was thinking about this about the weekend, I'l have to talk to the > debian group as well but I'l have an idea: > to be debian compliant AND to not interrupt with the way you provide your > package we could do: > > * Implement all packages debian specific (pathes, etc.) > * Create a freeswitch-pathcompatibility package which creates links to /opt > so scripts/users which are allready acustomed to your scheme > can have it their way, and the debianers out there which love their debian > conform system can simply deinstall / not install the pathcompatibility > package and everyone should be happy! > > What do you think of this idea? > > > i really would love to do this officially as the debian maintainer for > > the freeswitch project, i already sent the current packet builder an > > email but I did not receive an answer until know. > > > Feel free to toss patches at us via jira.freeswitch.org. We need to > start getting everything in place for 1.0.5. > > > Ok, i will doing my first adaptions starting this week. > > > I hope we can work together and that i will be accepted into this > > project which has in my oppinion a very bright future in the telecomm > > business! > > > Thanks. Also, feel free to catch up with us on irc to discuss any of > these issues. > > > I will do this, i am Hatrix (registered nick) in IRC, > > thanks, best > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/2ccbea9a/attachment.html From jerry.richards at teotech.com Mon Aug 24 12:26:29 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 24 Aug 2009 12:26:29 -0700 Subject: [Freeswitch-dev] Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED] Message-ID: <93866B3C643249C3AF0EC07206306946@greyhawk.tonecommander.com> Hello All, I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine for the first time using the Getting Started Guide. I can register three lines (1000, 1001, and 1002), but when I attempt to call one phone to the other I hear the operator say: "The person at extension 1000 is not available..." Also, the Freeswitch log shows: Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: [FACILITY_NOT_SUBSCRIBED] Does anyone know why I get this error? Best Regards, Jerry From prof at ukrpost.net Mon Aug 24 12:50:09 2009 From: prof at ukrpost.net (Sergey Popov) Date: Mon, 24 Aug 2009 22:50:09 +0300 Subject: [Freeswitch-dev] Strange NAT errors In-Reply-To: References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> Message-ID: <136816800.20090824225009@ukrpost.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/8a785926/attachment-0001.html From brian at freeswitch.org Mon Aug 24 13:08:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 24 Aug 2009 15:08:25 -0500 Subject: [Freeswitch-dev] Strange NAT errors In-Reply-To: <136816800.20090824225009@ukrpost.net> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> <136816800.20090824225009@ukrpost.net> Message-ID: <662A5BF2-4E1E-4F33-9E2E-6C49A6871418@freeswitch.org> We will need you to get a packet capture and open a jira... your uPNP router is doing something funky. The reason it didn't happen in March is we didn't have uPNP integrated... you can disable uPNP by starting FreeSWITCH with -nonat /b On Aug 24, 2009, at 2:50 PM, Sergey Popov wrote: > Hi, All! > > Last time FS on random time write NAT errors to console. It can > repeat oneself many times and stop randomly. After that occurrence > skypiax stop working. It occur on today version FS from SVN, and > binary installation 1.0.4. Version from March is free from this error. > > OS: Vista > Configuration: default + skypiax + mod_managed > IP address 192.168.0.2, NAT on router (D-Link DI-524) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/621e440b/attachment.html From anthony.minessale at gmail.com Mon Aug 24 13:17:00 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 24 Aug 2009 15:17:00 -0500 Subject: [Freeswitch-dev] Strange NAT errors In-Reply-To: <136816800.20090824225009@ukrpost.net> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> <136816800.20090824225009@ukrpost.net> Message-ID: <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> that looks like a bug to me in the code where uninit memory is confused for an ip change please try r14623 2009/8/24 Sergey Popov > Hi, All! > > Last time FS on random time write NAT errors to console. It can repeat > oneself many times and stop randomly. After that occurrence skypiax stop > working. It occur on today version FS from SVN, and binary installation > 1.0.4. Version from March is free from this error. > > OS: Vista > Configuration: default + skypiax + mod_managed > IP address 192.168.0.2, NAT on router (D-Link DI-524) > > Value of field network-address-change-v4 every time consist meaningless set > of random symbols. And my IP addresses is not changed really. What can it > be? > > 2009-08-24 18:20:50.910823 [INFO] switch_nat.c:305 Public IP changed from > '77.122.60.138' to '????????????????????h'. > 2009-08-24 18:20:50.910823 [INFO] switch_nat.c:392 Scanning for NAT > 2009-08-24 18:20:50.910823 [INFO] mod_sofia.c:3276 EVENT_TRAP: IP change > detected > 2009-08-24 18:20:50.910823 [INFO] mod_sofia.c:3288 IP change detected > [77.122.60.138]->[????????????????????h] []->[] > 2009-08-24 18:20:50.910823 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 > 2009-08-24 18:20:50.912823 [ERR] switch_nat.c:183 Error checking for PMP > [general error] > 2009-08-24 18:20:50.912823 [DEBUG] switch_nat.c:397 Checking for UPnP > 2009-08-24 18:20:51.775823 [NOTICE] sofia_glue.c:3506 Reload XML [Success] > 2009-08-24 18:20:51.775823 [INFO] mod_enum.c:808 ENUM Reloaded > 2009-08-24 18:20:51.775823 [INFO] switch_time.c:661 Timezone reloaded 530 > definitions > 2009-08-24 18:20:52.483823 [DEBUG] sofia.c:983 Write lock internal-ipv6 > 2009-08-24 18:20:52.497823 [NOTICE] sofia.c:990 Waiting for worker thread > 2009-08-24 18:20:52.498823 [DEBUG] sofia.c:1048 Write unlock internal-ipv6 > 2009-08-24 18:20:52.499823 [NOTICE] sofia.c:2850 Started Profile > internal-ipv6 [sofia_reg_internal-ipv6] > 2009-08-24 18:20:52.500823 [DEBUG] sofia.c:821 Creating agent for > internal-ipv6 > 2009-08-24 18:20:52.508823 [DEBUG] sofia.c:983 Write lock internal > 2009-08-24 18:20:52.509823 [DEBUG] sofia.c:983 Write lock external > 2009-08-24 18:20:52.523823 [NOTICE] sofia.c:990 Waiting for worker thread > 2009-08-24 18:20:52.527823 [DEBUG] sofia.c:1048 Write unlock internal > 2009-08-24 18:20:52.527823 [NOTICE] sofia.c:1517 Adding Alias [192.168.0.2] > for profile [internal] > 2009-08-24 18:20:52.527823 [NOTICE] sofia.c:2850 Started Profile internal > [sofia_reg_internal] > 2009-08-24 18:20:52.528823 [DEBUG] sofia.c:821 Creating agent for internal > 2009-08-24 18:20:52.541823 [NOTICE] sofia.c:990 Waiting for worker thread > 2009-08-24 18:20:52.545823 [NOTICE] sofia_glue.c:3567 deleted gateway > example.com > 2009-08-24 18:20:52.545823 [DEBUG] sofia.c:1048 Write unlock external > 2009-08-24 18:20:52.545823 [NOTICE] sofia_reg.c:2080 Added gateway ' > example.com' to profile 'external' > 2009-08-24 18:20:52.545823 [NOTICE] sofia.c:2850 Started Profile external > [sofia_reg_external] > 2009-08-24 18:20:52.545823 [DEBUG] sofia.c:821 Creating agent for external > 2009-08-24 18:20:52.607823 [DEBUG] sofia.c:876 Created agent for > internal-ipv6 > 2009-08-24 18:20:52.607823 [DEBUG] sofia.c:909 Set params for internal-ipv6 > 2009-08-24 18:20:52.607823 [DEBUG] sofia.c:929 Activated db for > internal-ipv6 > 2009-08-24 18:20:52.611823 [DEBUG] sofia.c:956 Starting thread for > internal-ipv6 > 2009-08-24 18:20:52.637823 [DEBUG] sofia.c:876 Created agent for external > 2009-08-24 18:20:52.637823 [DEBUG] sofia.c:909 Set params for external > 2009-08-24 18:20:52.637823 [DEBUG] sofia.c:929 Activated db for external > 2009-08-24 18:20:52.637823 [DEBUG] sofia.c:956 Starting thread for external > 2009-08-24 18:20:52.646823 [DEBUG] sofia.c:876 Created agent for internal > 2009-08-24 18:20:52.646823 [DEBUG] sofia.c:909 Set params for internal > 2009-08-24 18:20:52.646823 [DEBUG] sofia.c:929 Activated db for internal > 2009-08-24 18:20:52.657823 [DEBUG] sofia.c:956 Starting thread for internal > 2009-08-24 18:20:53.941823 [INFO] switch_nat.c:405 NAT detected type: upnp, > ExtIP: '77.122.60.138' > 2009-08-24 18:20:53.941823 [DEBUG] switch_nat.c:623 Refreshing nat maps > 2009-08-24 18:20:53.965823 [DEBUG] switch_nat.c:482 mapped public port 5060 > protocol UDP to localport 5060 > 2009-08-24 18:20:53.980823 [DEBUG] switch_nat.c:482 mapped public port 5060 > protocol TCP to localport 5060 > 2009-08-24 18:20:53.995823 [DEBUG] switch_nat.c:482 mapped public port 5080 > protocol UDP to localport 5080 > 2009-08-24 18:20:54.10823 [DEBUG] switch_nat.c:482 mapped public port 5080 > protocol TCP to localport 5080 > 2009-08-24 18:20:54.10823 [INFO] switch_nat.c:305 Public IP changed from > '77.122.60.138' to '????????????????????h'. > 2009-08-24 18:20:54.10823 [INFO] mod_sofia.c:3276 EVENT_TRAP: IP change > detected > 2009-08-24 18:20:54.10823 [INFO] mod_sofia.c:3288 IP change detected > [77.122.60.138]->[????????????????????h] []->[] > 2009-08-24 18:20:54.10823 [INFO] switch_nat.c:392 Scanning for NAT > 2009-08-24 18:20:54.10823 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 > 2009-08-24 18:20:54.13823 [ERR] switch_nat.c:183 Error checking for PMP > [general error] > 2009-08-24 18:20:54.13823 [DEBUG] switch_nat.c:397 Checking for UPnP > 2009-08-24 18:20:54.847823 [NOTICE] sofia_glue.c:3506 Reload XML [Success] > 2009-08-24 18:20:54.847823 [ERR] sofia_glue.c:3522 Profile internal must be > up for at least 10 seconds to stop/restart. > Please wait 8 seconds > 2009-08-24 18:20:54.847823 [ERR] sofia_glue.c:3522 Profile external must be > up for at least 10 seconds to stop/restart. > Please wait 8 seconds > 2009-08-24 18:20:54.847823 [ERR] sofia_glue.c:3522 Profile internal-ipv6 > must be up for at least 10 seconds to stop/restart. > Please wait 8 seconds > 2009-08-24 18:20:54.847823 [ERR] sofia_glue.c:3522 Profile internal must be > up for at least 10 seconds to stop/restart. > Please wait 8 seconds > 2009-08-24 18:20:54.847823 [INFO] mod_enum.c:808 ENUM Reloaded > 2009-08-24 18:20:54.848823 [INFO] switch_time.c:661 Timezone reloaded 530 > definitions > 2009-08-24 18:20:57.40823 [INFO] switch_nat.c:405 NAT detected type: upnp, > ExtIP: '77.122.60.138' > 2009-08-24 18:20:57.40823 [DEBUG] switch_nat.c:623 Refreshing nat maps > 2009-08-24 18:20:57.61823 [DEBUG] switch_nat.c:482 mapped public port 5060 > protocol UDP to localport 5060 > 2009-08-24 18:20:57.75823 [DEBUG] switch_nat.c:482 mapped public port 5060 > protocol TCP to localport 5060 > 2009-08-24 18:20:57.88823 [DEBUG] switch_nat.c:482 mapped public port 5080 > protocol UDP to localport 5080 > 2009-08-24 18:20:57.102823 [DEBUG] switch_nat.c:482 mapped public port 5080 > protocol TCP to localport 5080 > 2009-08-24 18:20:57.102823 [INFO] switch_nat.c:305 Public IP changed from > '77.122.60.138' to '????????????????????h'. > 2009-08-24 18:20:57.102823 [INFO] switch_nat.c:392 Scanning for NAT > 2009-08-24 18:20:57.102823 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 > 2009-08-24 18:20:57.103823 [INFO] mod_sofia.c:3276 EVENT_TRAP: IP change > detected > 2009-08-24 18:20:57.103823 [INFO] mod_sofia.c:3288 IP change detected > [77.122.60.138]->[????????????????????h] []->[] > 2009-08-24 18:20:57.103823 [ERR] switch_nat.c:183 Error checking for PMP > [general error] > 2009-08-24 18:20:57.103823 [DEBUG] switch_nat.c:397 Checking for UPnP > 2009-08-24 18:20:57.992823 [NOTICE] sofia_glue.c:3506 Reload XML [Success] > 2009-08-24 18:20:57.992823 [ERR] sofia_glue.c:3522 Profile internal must be > up for at least 10 seconds to stop/restart. > Please wait 5 seconds > 2009-08-24 18:20:57.992823 [ERR] sofia_glue.c:3522 Profile external must be > up for at least 10 seconds to stop/restart. > Please wait 5 seconds > 2009-08-24 18:20:57.993823 [ERR] sofia_glue.c:3522 Profile internal-ipv6 > must be up for at least 10 seconds to stop/restart. > Please wait 5 seconds > 2009-08-24 18:20:57.993823 [ERR] sofia_glue.c:3522 Profile internal must be > up for at least 10 seconds to stop/restart. > Please wait 5 seconds > 2009-08-24 18:20:57.993823 [INFO] mod_enum.c:808 ENUM Reloaded > 2009-08-24 18:20:57.993823 [INFO] switch_time.c:661 Timezone reloaded 530 > definitions > > Event-Name: TRAP > Core-UUID: f1764d44-a470-834c-9a07-c0225a65232d > FreeSWITCH-Hostname: ProfNest > FreeSWITCH-IPv4: 192.168.0.2 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-08-24%2018%3A22%3A54 > Event-Date-GMT: Mon,%2024%20Aug%202009%2015%3A22%3A54%20GMT > Event-Date-Timestamp: 1251127374637823 > Event-Calling-File: switch_nat.c > Event-Calling-Function: switch_nat_multicast_runtime > Event-Calling-Line-Number: 308 > condition: network-address-change > network-address-previous-v4: 77.122.60.138 > network-address-change-v4: > %?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?C%?Ch > > > Bye. > > Monday, August 24, 2009, 18:40 Sergey Popov. > --- > * E-mail prof at ukrpost.net * ICQ 2831794 * > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090824/b0295851/attachment.html From janvb at live.com Tue Aug 25 03:33:57 2009 From: janvb at live.com (Jan Berger) Date: Tue, 25 Aug 2009 12:33:57 +0200 Subject: [Freeswitch-dev] SS7/SIGTRAN stack In-Reply-To: <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> <136816800.20090824225009@ukrpost.net> <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> Message-ID: hi, Attaching some work on M2UA if someone is interested in reviewing/commenting the work. I have started the long path to create a full SS7 and SIGTRAN stack, and the way I intend to do this work is by using a code generator I am developing myself. I started with M2UA as a test case, and you will find the xml spec + generated MAUP encoder/decoder attached (I have only set of the MAUP part of M2UA as a test case). My first objective here is to complete the code generator so it generate encoders/decoders for any protocol. This alone removes a bit load of work and is a proven technique on writing protocols. What's new with this generator is that it uses a detailed xml spec allowing the encoder/decoder to be generated 100% + it can support both ASN.1 and non-ASN.1 protocols. I am currently using M2UA as a test case, but have also a test case for MTP and ISUP. The second objective is to create M2UA, (maybe M2PA), MTP2, MTP3 and ISUP using this tool, and bring this stack to a level where we can do a simple IAM based call in/out. Comments/suggestions are welcome. Jan _________________________________________________________________ More than messages?check out the rest of the Windows Live?. http://www.microsoft.com/windows/windowslive/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/1dfd6a4b/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: m2ua.xml Type: text/xml Size: 5894 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/1dfd6a4b/attachment-0001.xml -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: m2ua_mes.h Url: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/1dfd6a4b/attachment-0001.h -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: m2ua_mes.c Url: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/1dfd6a4b/attachment-0001.c From wasim at convergence.pk Tue Aug 25 03:53:59 2009 From: wasim at convergence.pk (Wasim Baig) Date: Tue, 25 Aug 2009 15:53:59 +0500 Subject: [Freeswitch-dev] SS7/SIGTRAN stack In-Reply-To: References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> <136816800.20090824225009@ukrpost.net> <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> Message-ID: On Tue, Aug 25, 2009 at 3:33 PM, Jan Berger wrote: Attaching some work on M2UA if someone is interested in reviewing/commenting > the work. > Excellent. Sorely needed. I have access to a number of test beds that can MTP and Sigtran. Count me in this effort. > The second objective is to create M2UA, (maybe M2PA), MTP2, MTP3 and ISUP > using this tool, and bring this stack to a level where we can do a simple > IAM based call in/out. > Hallelujah. -- wasim h. baig | principal consultant | convergence pk | +92 300 8508070 | peace be upon you ... Sent from Lahore, Punjab, Pakistan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/1a09b6bc/attachment.html From prof at ukrpost.net Tue Aug 25 05:44:48 2009 From: prof at ukrpost.net (Sergey Popov) Date: Tue, 25 Aug 2009 15:44:48 +0300 Subject: [Freeswitch-dev] Strange NAT errors In-Reply-To: <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> References: <3F95FE6D-D9E3-4ABF-9BB6-E0CFE17835F3@runsolutions.com> <868B9A5A-DB64-4EC8-B98E-0FE71AC7C4D8@jerris.com> <3181A30B8C35AB4AA8577B78DDF4613805A2ADE8@nickel.mettonigroup.com> <136816800.20090824225009@ukrpost.net> <191c3a030908241317u16428356of12c4c10e0399f@mail.gmail.com> Message-ID: <908679758.20090825154448@ukrpost.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090825/ee49c3a9/attachment.html From csa at nowthor.com Tue Aug 25 07:06:25 2009 From: csa at nowthor.com (Carlos S. Antunes) Date: Tue, 25 Aug 2009 10:06:25 -0400 Subject: [Freeswitch-dev] [Freeswitch-users] how to avoid many "|" in bridge application? In-Reply-To: References: Message-ID: <4A93EFE1.6010007@nowthor.com> Max, I would like to see something similar too. For example, it would be wonderful if one could specify multiple gateways to try like this or something similar: One would be able to avoid the "[]" and "{}" hacks and combine sequential and simultaneous trying of gateways. What do the developers think of this? Carlos Max Ivanov wrote: > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie > sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000 > , > the whole sting is tooo long, is there any shorter way to write same thing? Like > "sofia/gateway/panas*/1000" will try all gateways matching the pattern. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jerry.richards at teotech.com Wed Aug 26 13:15:12 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 26 Aug 2009 13:15:12 -0700 Subject: [Freeswitch-dev] Scalabilty of Freeswitch Message-ID: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Best Regards, Jerry From anthony.minessale at gmail.com Wed Aug 26 13:30:55 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 26 Aug 2009 15:30:55 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> Message-ID: <191c3a030908261330j75037b17l1d87c082bdf7c89d@mail.gmail.com> You might want to just ask the user list for their experiences. We try not to address load and scale questions directly to avoid a conflict of interest. There are many ways to cluster FS but I don't have time to explain it now, you can find lots of information on the wiki and by asking the users list and IRC I can say a top-of-the-line server can at least do call volume measured in thousands not hundreds of call legs. On Wed, Aug 26, 2009 at 3:15 PM, Jerry Richards wrote: > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > > Also, when that number is exceeded, how can Freeswitch server be > distributed > to accommodate a larger installation? > > Best Regards, > Jerry > > > _______________________________________________ > 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090826/e618d7cf/attachment.html From sramsey at sipinterchange.com Wed Aug 26 13:31:27 2009 From: sramsey at sipinterchange.com (Shelby Ramsey) Date: Wed, 26 Aug 2009 15:31:27 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> Message-ID: <4A959B9F.8000308@sipinterchange.com> Jerry, This is a pretty generic question. Generically speaking users aren't what most people think of but some good guidelines are: -- 2000 simultaneous calls (probably more depending on your CPS) if you are not running any media through the box -- I'm not sure with media what the upper end would be ... but I would seriously doubt you'll get acceptable performance of more than 500 - 700 calls on a single interface OpenSIPs works well as a load balancer. As for FS itself there is no built in mechanism to share state across multiple FS instances ... however you can do it yourself with mod_event_socket. SDR Jerry Richards wrote: > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > > Also, when that number is exceeded, how can Freeswitch server be distributed > to accommodate a larger installation? > > Best Regards, > Jerry > > > _______________________________________________ > 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 msc at freeswitch.org Wed Aug 26 13:35:44 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 26 Aug 2009 13:35:44 -0700 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> Message-ID: <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. > > Also, when that number is exceeded, how can Freeswitch server be > distributed > to accommodate a larger installation? > Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090826/7aaab258/attachment.html From mkezys at gmail.com Wed Aug 26 22:37:37 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Thu, 27 Aug 2009 08:37:37 +0300 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> Message-ID: <07d201ca26d8$7d4a8e70$77dfab50$@com> Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090827/2c491d08/attachment.html From mike at jerris.com Thu Aug 27 07:44:20 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 27 Aug 2009 10:44:20 -0400 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <07d201ca26d8$7d4a8e70$77dfab50$@com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> Message-ID: <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: > Maybe it would be a good idea to create wiki page just to put such > kind of information in a table: > > Computer specs | Other comments | Codecs used | With/Without Media | > Max sim. calls reached | etc > > That way interested persons could get a grasp what is really all > about. > > This is very common question based on which many people measure > switch capabilities, so in my opinion should be treated with that in > mind (as marketing oportunity) > > Regards, > Mindaugas Kezys > http://www.kolmisoft.com > VoIP Billing and Routing Solutions > > From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch- > dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: 2009 m. rugpj??io 26 d. 23:36 > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch > > This question sounds eerily familiar... > > On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards > wrote: > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch > (assuming a > top-of-the-line server), in terms of how many users? > > You can have many hundreds of users, but there are a lot of factors: > network infrastructure, call volume, etc. > > > Also, when that number is exceeded, how can Freeswitch server be > distributed > to accommodate a larger installation? > > Yes there are strategies. You definitely want a professional to > assist if this is a serious production environment. There are > members of the FS community who do this sort of thing, or you could > email consulting at freeswitch.org to get assistance from the core FS > developers. > -MC > > _______________________________________________ > 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/20090827/57f0b64d/attachment.html From mkezys at gmail.com Thu Aug 27 07:56:40 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Thu, 27 Aug 2009 17:56:40 +0300 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> Message-ID: <09a001ca2726$9664a820$c32df860$@com> I do not use Freeswitch currently, but monitoring closely. Will start developing on Freeswitch after few months and will do tests like this for sure. If nobody will create such page ? I will definitely, but not now ? because have not such info. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: 2009 m. rugpj??io 27 d. 17:44 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _______________________________________________ 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/20090827/8c302eaf/attachment-0001.html From jerry.richards at teotech.com Thu Aug 27 08:45:30 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 27 Aug 2009 08:45:30 -0700 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <07d201ca26d8$7d4a8e70$77dfab50$@com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com><87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> Message-ID: <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> Also on a related issue (since I am a novice to Freeswitch), regardless of statistics on number of extensions per system per configuration, I had a more general question: That is, does Freeswitch allow for distribution of loading (e.g. supports interface to media servers running on separate machines)? Also, I imagine the SIP signaling part must always be a standalone machine that manages all extensions? Best Regards, Jerry _____ From: Mindaugas Kezys [mailto:mkezys at gmail.com] Sent: Wednesday, August 26, 2009 10:38 PM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090827/7b76a9e3/attachment.html From sramsey at sipinterchange.com Thu Aug 27 09:04:49 2009 From: sramsey at sipinterchange.com (Shelby Ramsey) Date: Thu, 27 Aug 2009 11:04:49 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com><87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> Message-ID: <4A96AEA1.7050701@sipinterchange.com> Jerry, FS does not work in the manner that you are thinking. Your option is to run the signaling alone and not pass the media OR to pass the media. You can not have signaling run from one FS instance and have it pass media through another instance (unless you were to "route" the traffic to the other instance via SIP or another endpoint). Many architectures (in the commercial carrier grade products) typically separate functionality into different processes and use some sort of network based IPC: -- the call processing -- the endpoints (i.e. SIP, H323) ... which use IPC to talk to the call processing unit -- the media server ... which are controlled by IPC by the call processing unit But in the case of FS this is not possible. A single FS process lives in it's own world (although it can be quite verbose in telling you what you want to know) with no knowledge of other FS processes. Many people have been quite successful using things like event_socket to load balance calls across a cluster of FS instances (which can look the same as the above). I'm sure other people on the user list can give you great suggestions if you ask what you are trying to accomplish (I'd suggest the general list). The community is pretty generous. Hope this helps. SDR Jerry Richards wrote: > Also on a related issue (since I am a novice to > Freeswitch), regardless of statistics on number of extensions per > system per configuration, I had a more general question: > > That is, does Freeswitch allow for distribution of loading (e.g. > supports interface to media servers running on separate machines)? > Also, I imagine the SIP signaling part must always be a standalone > machine that manages all extensions? > > Best Regards, > Jerry > > > ------------------------------------------------------------------------ > *From:* Mindaugas Kezys [mailto:mkezys at gmail.com] > *Sent:* Wednesday, August 26, 2009 10:38 PM > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > Maybe it would be a good idea to create wiki page just to put such > kind of information in a table: > > > > Computer specs | Other comments | Codecs used | With/Without Media > | Max sim. calls reached | etc > > > > That way interested persons could get a grasp what is really all > about. > > > > This is very common question based on which many people measure > switch capabilities, so in my opinion should be treated with that > in mind (as marketing oportunity) > > > > Regards, > > Mindaugas Kezys > > http://www.kolmisoft.com > > VoIP Billing and Routing Solutions > > > > *From:* freeswitch-dev-bounces at lists.freeswitch.org > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of > *Michael Collins > *Sent:* 2009 m. rugpj??io 26 d. 23:36 > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > This question sounds eerily familiar... > > On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards > > > wrote: > > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch > (assuming a > top-of-the-line server), in terms of how many users? > > > You can have many hundreds of users, but there are a lot of > factors: network infrastructure, call volume, etc. > > > > Also, when that number is exceeded, how can Freeswitch server > be distributed > to accommodate a larger installation? > > > Yes there are strategies. You definitely want a professional to > assist if this is a serious production environment. There are > members of the FS community who do this sort of thing, or you > could email consulting at freeswitch.org > to get assistance from the core > FS developers. > -MC > > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 krice at freeswitch.org Thu Aug 27 09:31:01 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 27 Aug 2009 11:31:01 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> Message-ID: It is possible to do such a think via a number of mechanisms. Remember FS can operate in 3 modes, standard mode where we can do transcoding and interact with the media stream, or proxy_media mode where we just proxy the media packet in packet out, and bypass_media mode where we tell the end points to send the media directly to each other. This allows for some interesting scalability (we have clusters running in excess of 20K concurrent calls at call rates in excess of 2000 calls/sec) From: Jerry Richards Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Thu, 27 Aug 2009 08:45:30 -0700 To: Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch Also on a related issue (since I am a novice to Freeswitch), regardless of statistics on number of extensions per system per configuration, I had a more general question: That is, does Freeswitch allow for distribution of loading (e.g. supports interface to media servers running on separate machines)? Also, I imagine the SIP signaling part must always be a standalone machine that manages all extensions? Best Regards, Jerry > > > > From: Mindaugas Kezys [mailto:mkezys at gmail.com] > Sent: Wednesday, August 26, 2009 10:38 PM > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > > > Maybe it would be a good idea to create wiki page just to put such kind of > information in a table: > > > > Computer specs | Other comments | Codecs used | With/Without Media | Max sim. > calls reached | etc > > > > That way interested persons could get a grasp what is really all about. > > > > This is very common question based on which many people measure switch > capabilities, so in my opinion should be treated with that in mind (as > marketing oportunity) > > > > Regards, > > Mindaugas Kezys > > http://www.kolmisoft.com > > VoIP Billing and Routing Solutions > > > > > > From: freeswitch-dev-bounces at lists.freeswitch.org > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: 2009 m. rugpj??io 26 d. 23:36 > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > This question sounds eerily familiar... > > > > On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards > wrote: > > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > > > > > You can have many hundreds of users, but there are a lot of factors: network > infrastructure, call volume, etc. > > >> >> >> Also, when that number is exceeded, how can Freeswitch server be distributed >> to accommodate a larger installation? > > > > > Yes there are strategies. You definitely want a professional to assist if > this is a serious production environment. There are members of the FS > community who do this sort of thing, or you could email > consulting at freeswitch.org to get assistance from the core FS developers. > -MC > > > > > _______________________________________________ > 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/20090827/42e37422/attachment-0001.html From mgg at giagnocavo.net Thu Aug 27 09:32:44 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 27 Aug 2009 12:32:44 -0400 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702D7CC237E@mse17be1.mse17.exchange.ms> Can?t we do what some vendors do? Pick the simplest config for a simple scenario, like statically bridging two channels, then publish those numbers? Or say, use one of the standard SIPP scenarios with no RTP? And so on. From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, August 27, 2009 8:44 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards > wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _______________________________________________ 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/20090827/a98bf4ce/attachment.html From anthony.minessale at gmail.com Thu Aug 27 10:07:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 27 Aug 2009 12:07:01 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702D7CC237E@mse17be1.mse17.exchange.ms> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> <6E8D2069C08AA84A83D336E996AE4C6702D7CC237E@mse17be1.mse17.exchange.ms> Message-ID: <191c3a030908271007k6523b39fqc4b32fbf335f521b@mail.gmail.com> I think there are 3 main problems. 1) The core developers should not be blowing their own horn about performance results in any official capacity because it will appear skewed to skeptics. 2) The core developers are too busy to bother with gathering the stats. 3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc. It sort of like if someone offered FREE CARS, take one and drive away and do whatever you want with it. People show up and say, "how fast can it go?" "how well does it handle on curves?" .. I don't know it's a FREE CAR, go drive it and you tell me. I think some people forget this is an open source project and it's up to the community to decide for themselves how well it works. I would love to find a way to save us all the time of dealing with people who try to load test first and try real calls later. The industry standard for calls is 50cps which is assuming a typical load of 25cps burst-able to 50 to cover a fault in a cluster. Most people doing 50cps all day long should be making so much money they can either afford nicer boxes or more expensive equipment if they need to scale more. FS can handle this load easily and many people tune their box to do triple that if not even more but that is all subjective to their hardware choices etc. sigh, On Thu, Aug 27, 2009 at 11:32 AM, Michael Giagnocavo wrote: > Can?t we do what some vendors do? Pick the simplest config for a simple > scenario, like statically bridging two channels, then publish those numbers? > Or say, use one of the standard SIPP scenarios with no RTP? And so on. > > > > *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Michael Jerris > *Sent:* Thursday, August 27, 2009 8:44 AM > > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > if you have the time and ability to create such a multi-dementional array > of tjat data in some usable form that would be fine, it is a wiki after all. > I however tend to think that there are too many variables to reliably > provide any sort of real data and that the time required to do so would be > quite a lot. > > > > Mike > > > > On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: > > > > Maybe it would be a good idea to create wiki page just to put such kind > of information in a table: > > > > Computer specs | Other comments | Codecs used | With/Without Media | Max > sim. calls reached | etc > > > > That way interested persons could get a grasp what is really all about. > > > > This is very common question based on which many people measure switch > capabilities, so in my opinion should be treated with that in mind (as > marketing oportunity) > > > > Regards, > > Mindaugas Kezys > > http://www.kolmisoft.com > > VoIP Billing and Routing Solutions > > > > *From:* freeswitch-dev-bounces at lists.freeswitch.org [mailto: > freeswitch-dev-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* 2009 m. rugpj??io 26 d. 23:36 > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > This question sounds eerily familiar... > > On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards < > jerry.richards at teotech.com> wrote: > > Hello All, > > Does anyone know what the capacity of a stand-alone Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > > > You can have many hundreds of users, but there are a lot of factors: > network infrastructure, call volume, etc. > > > > Also, when that number is exceeded, how can Freeswitch server be > distributed > to accommodate a larger installation? > > > Yes there are strategies. You definitely want a professional to assist if > this is a serious production environment. There are members of the FS > community who do this sort of thing, or you could email > consulting at freeswitch.org to get assistance from the core FS developers. > -MC > > > > _______________________________________________ > 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 > > -- 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090827/80a2ceca/attachment-0001.html From anatoliy at kounitskiy.com Thu Aug 27 13:18:43 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Thu, 27 Aug 2009 23:18:43 +0300 Subject: [Freeswitch-dev] Unable to access directory variable from channel In-Reply-To: <27c25bc40908200550h27bf96dfrf0ba1833ca0619b9@mail.gmail.com> References: <27c25bc40908200550h27bf96dfrf0ba1833ca0619b9@mail.gmail.com> Message-ID: <1cd828b60908271318n420ca2e9u16e1fc7ec681cd4d@mail.gmail.com> You can try also application/function user_data http://wiki.freeswitch.org/wiki/Mod_commands#user_data Also here ( http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Alphanumeric_to_numeric_user_mapping ) you can see it in use with application bridge. Works perfectly with set/export applications. Anatoliy On Thu, Aug 20, 2009 at 3:50 PM, Juan Backson wrote: > Hi, > > I set up a variable called "account-id" in the user directory xml file. > > From the info app, I can't find the variable for variable_account-id. > Also,? in my C module, I tried to obtain the value with > switch_channel_get_variable(channel,"account-id") and it is not returning > null. > > Does anyone know how I can access the user variable from within my mod? > > Thanks, > JB > _______________________________________________ > 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 > > -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From jerry.richards at teotech.com Thu Aug 27 14:03:29 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 27 Aug 2009 14:03:29 -0700 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: References: <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> Message-ID: <4AC67E228CB24A3D9D6A206F46CD1582@greyhawk.tonecommander.com> Okay. I got two replies from Shelby R. and Ken R. I presume these two answers do not conflict? I gather from these replies is that FS has 3 modes: 1) FS is a complete stand-alone system (i.e. FS interacts with media stream), 2) FS is a media_proxy (i.e. FS simply receives/transmits media but does not process it), 3) FS is in bypass media mode where FS tells both endpoints to send media directly to each other (i.e. no media handling packet handling at all). In mode 3), How do the IVR, Voice Mail, Eavesdropping, MOH and central conferencing features work? Would it use an external media server(s)? Also, Ken you mentioned "clusters". What is a cluster? What system(s) are you running the 20K concurrent calls at 2000 calls/sec rate? Which of the three modes are you running the server when recording these statistics? Thanks and Best Regards, Jerry _____ From: Ken Rice [mailto:krice at freeswitch.org] Sent: Thursday, August 27, 2009 9:31 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch It is possible to do such a think via a number of mechanisms. Remember FS can operate in 3 modes, standard mode where we can do transcoding and interact with the media stream, or proxy_media mode where we just proxy the media packet in packet out, and bypass_media mode where we tell the end points to send the media directly to each other. This allows for some interesting scalability (we have clusters running in excess of 20K concurrent calls at call rates in excess of 2000 calls/sec) _____ From: Jerry Richards Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Thu, 27 Aug 2009 08:45:30 -0700 To: Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch Also on a related issue (since I am a novice to Freeswitch), regardless of statistics on number of extensions per system per configuration, I had a more general question: That is, does Freeswitch allow for distribution of loading (e.g. supports interface to media servers running on separate machines)? Also, I imagine the SIP signaling part must always be a standalone machine that manages all extensions? Best Regards, Jerry _____ From: Mindaugas Kezys [mailto:mkezys at gmail.com] Sent: Wednesday, August 26, 2009 10:38 PM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _____ _______________________________________________ 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/20090827/1f3bb99a/attachment.html From krice at freeswitch.org Thu Aug 27 14:23:31 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 27 Aug 2009 16:23:31 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <4AC67E228CB24A3D9D6A206F46CD1582@greyhawk.tonecommander.com> Message-ID: In mode 3 FS does not handle the media for any of those things you have to depend on a secondary system running FS or some other platform to do those things. Even in mode 3 FS is not a SIP proxy it is still a B2BUA but does nothing with the media simply directs it to where you want it handled When I mentioned clustering it together I was speaking strictly horizontal scalability using a DB to share information between the servers such as registered end points etc... This be accomplished in several different ways it really depends on what exactly you are trying to accomplish. For that particular setup it was a very large call center with dozens of SIP aware machines behind the FS boxes handling the IVRs, call recording and call distribution. Ken From: Jerry Richards Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Thu, 27 Aug 2009 14:03:29 -0700 To: Cc: Steve Hill , Joe Billey , Richard Lee , Thomas Beck Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch Okay. I got two replies from Shelby R. and Ken R. I presume these two answers do not conflict? I gather from these replies is that FS has 3 modes: 1) FS is a complete stand-alone system (i.e. FS interacts with media stream), 2) FS is a media_proxy (i.e. FS simply receives/transmits media but does not process it), 3) FS is in bypass media mode where FS tells both endpoints to send media directly to each other (i.e. no media handling packet handling at all). In mode 3), How do the IVR, Voice Mail, Eavesdropping, MOH and central conferencing features work? Would it use an external media server(s)? Also, Ken you mentioned "clusters". What is a cluster? What system(s) are you running the 20K concurrent calls at 2000 calls/sec rate? Which of the three modes are you running the server when recording these statistics? Thanks and Best Regards, Jerry > > > > From: Ken Rice [mailto:krice at freeswitch.org] > Sent: Thursday, August 27, 2009 9:31 AM > To: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > It is possible to do such a think via a number of mechanisms. Remember FS can > operate in 3 modes, standard mode where we can do transcoding and interact > with the media stream, or proxy_media mode where we just proxy the media > packet in packet out, and bypass_media mode where we tell the end points to > send the media directly to each other. > > This allows for some interesting scalability (we have clusters running in > excess of 20K concurrent calls at call rates in excess of 2000 calls/sec) > > > > > > From: Jerry Richards > Reply-To: "freeswitch-dev at lists.freeswitch.org" > > Date: Thu, 27 Aug 2009 08:45:30 -0700 > To: > Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch > > Also on a related issue (since I am a novice to Freeswitch), regardless of > statistics on number of extensions per system per configuration, I had a more > general question: > > That is, does Freeswitch allow for distribution of loading (e.g. supports > interface to media servers running on separate machines)? Also, I imagine > the SIP signaling part must always be a standalone machine that manages all > extensions? > > Best Regards, > Jerry > > > >> >> >> >> >> From: Mindaugas Kezys [mailto:mkezys at gmail.com] >> Sent: Wednesday, August 26, 2009 10:38 PM >> To: freeswitch-dev at lists.freeswitch.org >> Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch >> >> >> >> >> >> Maybe it would be a good idea to create wiki page just to put such kind of >> information in a table: >> >> >> >> Computer specs | Other comments | Codecs used | With/Without Media | Max >> sim. calls reached | etc >> >> >> >> That way interested persons could get a grasp what is really all about. >> >> >> >> This is very common question based on which many people measure switch >> capabilities, so in my opinion should be treated with that in mind (as >> marketing oportunity) >> >> >> >> Regards, >> >> Mindaugas Kezys >> >> http://www.kolmisoft.com >> >> VoIP Billing and Routing Solutions >> >> >> >> >> >> From: freeswitch-dev-bounces at lists.freeswitch.org >> [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael >> Collins >> Sent: 2009 m. rugpj??io 26 d. 23:36 >> To: freeswitch-dev at lists.freeswitch.org >> Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch >> >> >> >> This question sounds eerily familiar... >> >> >> >> On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards >> wrote: >> >> Hello All, >> >> Does anyone know what the capacity of a stand-alone Freeswitch (assuming a >> top-of-the-line server), in terms of how many users? >> >> >> >> >> You can have many hundreds of users, but there are a lot of factors: >> network infrastructure, call volume, etc. >> >> >> >>> >>> >>> Also, when that number is exceeded, how can Freeswitch server be >>> distributed >>> to accommodate a larger installation? >> >> >> >> >> Yes there are strategies. You definitely want a professional to assist if >> this is a serious production environment. There are members of the FS >> community who do this sort of thing, or you could email >> consulting at freeswitch.org to get assistance from the core FS developers. >> -MC >> >> >> >> >> >> >> _______________________________________________ >> 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/20090827/9ab8f765/attachment-0001.html From sramsey at sipinterchange.com Thu Aug 27 14:31:05 2009 From: sramsey at sipinterchange.com (Shelby Ramsey) Date: Thu, 27 Aug 2009 16:31:05 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <4AC67E228CB24A3D9D6A206F46CD1582@greyhawk.tonecommander.com> References: <627599E9CA3C445AB7D82EF17E27CF65@greyhawk.tonecommander.com> <4AC67E228CB24A3D9D6A206F46CD1582@greyhawk.tonecommander.com> Message-ID: <4A96FB19.10407@sipinterchange.com> Jerry, Our responses did not conflict at all (although Ken is definitely a more reliable resource on the topic than myself). To answer your question(s) (I think): -- The media modes really dictate how much traffic you can run through an instance -- If you select bypass media you can only control the signaling ... things like voicemail, etc would be managed by the devices you are signaling to (i.e. other FS instances). Most everyone who runs lots of traffic through FS clusters the devices. These clusters are independent FS instances that have no knowledge of one another. There are lots of ways to build your application so that they look like one (event_socket, xml_curl, custom mods) .... but FS itself won't take care of keeping the state from all of the devices in the cluster (nor will one instance tell another instance to eavesdrop on a call). It's actually pretty easy to build your own (thanks to the smarts these fine folks used when writing this thing). OpenSER(or any of the other forks of the project) is a great mechanism to load balance to the cluster or you could do networking tricks like round robin DNS. If you really wanted to dig into this I'd post what you are trying to accomplish on the list (general -- not dev) or if there are commercial concerns you could always engage the core developers in evaluating -- consulting at freeswitch.org. SDR Jerry Richards wrote: > Okay. I got two replies from Shelby R. and Ken R. I presume these > two answers do not conflict? I gather from these replies is that FS > has 3 modes: 1) FS is a complete stand-alone system (i.e. FS interacts > with media stream), 2) FS is a media_proxy (i.e. FS simply > receives/transmits media but does not process it), 3) FS is in bypass > media mode where FS tells both endpoints to send media directly to > each other (i.e. no media handling packet handling at all). > > In mode 3), How do the IVR, Voice Mail, Eavesdropping, MOH and central > conferencing features work? Would it use an external media server(s)? > > Also, Ken you mentioned "clusters". What is a cluster? What > system(s) are you running the 20K concurrent calls at 2000 calls/sec > rate? Which of the three modes are you running the server when > recording these statistics? > > Thanks and Best Regards, > Jerry > > ------------------------------------------------------------------------ > *From:* Ken Rice [mailto:krice at freeswitch.org] > *Sent:* Thursday, August 27, 2009 9:31 AM > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > It is possible to do such a think via a number of mechanisms. > Remember FS can operate in 3 modes, standard mode where we can do > transcoding and interact with the media stream, or proxy_media > mode where we just proxy the media packet in packet out, and > bypass_media mode where we tell the end points to send the media > directly to each other. > > This allows for some interesting scalability (we have clusters > running in excess of 20K concurrent calls at call rates in excess > of 2000 calls/sec) > > > > ------------------------------------------------------------------------ > *From: *Jerry Richards > *Reply-To: *"freeswitch-dev at lists.freeswitch.org" > > *Date: *Thu, 27 Aug 2009 08:45:30 -0700 > *To: * > *Subject: *Re: [Freeswitch-dev] Scalabilty of Freeswitch > > Also on a related issue (since I am a novice to Freeswitch), > regardless of statistics on number of extensions per system per > configuration, I had a more general question: > > That is, does Freeswitch allow for distribution of loading (e.g. > supports interface to media servers running on separate machines)? > Also, I imagine the SIP signaling part must always be a > standalone machine that manages all extensions? > > Best Regards, > Jerry > > > > > ------------------------------------------------------------------------ > *From:* Mindaugas Kezys [mailto:mkezys at gmail.com] > > *Sent:* Wednesday, August 26, 2009 10:38 PM > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > > > Maybe it would be a good idea to create wiki page just to put > such kind of information in a table: > > > > Computer specs | Other comments | Codecs used | With/Without > Media | Max sim. calls reached | etc > > > > That way interested persons could get a grasp what is really > all about. > > > > This is very common question based on which many people > measure switch capabilities, so in my opinion should be > treated with that in mind (as marketing oportunity) > > > > Regards, > > Mindaugas Kezys > > http://www.kolmisoft.com > > VoIP Billing and Routing Solutions > > > > > > *From:* freeswitch-dev-bounces at lists.freeswitch.org > [mailto:freeswitch-dev-bounces at lists.freeswitch.org] > *On > Behalf Of *Michael Collins > *Sent:* 2009 m. rugpj??io 26 d. 23:36 > *To:* freeswitch-dev at lists.freeswitch.org > *Subject:* Re: [Freeswitch-dev] Scalabilty of Freeswitch > > > > This question sounds eerily familiar... > > > > On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards > wrote: > > Hello All, > > Does anyone know what the capacity of a stand-alone > Freeswitch (assuming a > top-of-the-line server), in terms of how many users? > > > > > You can have many hundreds of users, but there are a lot of > factors: network infrastructure, call volume, etc. > > > > > > Also, when that number is exceeded, how can Freeswitch > server be distributed > to accommodate a larger installation? > > > > > > Yes there are strategies. You definitely want a professional > to assist if this is a serious production environment. There > are members of the FS community who do this sort of thing, or > you could email consulting at freeswitch.org to get assistance > from the core FS developers. > -MC > > > > ------------------------------------------------------------------------ > > _______________________________________________ > 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 mkezys at gmail.com Thu Aug 27 22:55:20 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Fri, 28 Aug 2009 08:55:20 +0300 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <191c3a030908271007k6523b39fqc4b32fbf335f521b@mail.gmail.com> References: <1B5F9FFF4AD7487BB31A862A6FE379E4@greyhawk.tonecommander.com> <87f2f3b90908261335p4a1f6c7k6a448d7d9d4dad94@mail.gmail.com> <07d201ca26d8$7d4a8e70$77dfab50$@com> <60FDDD7A-D9AC-45CE-9B7E-FF122161A69F@jerris.com> <6E8D2069C08AA84A83D336E996AE4C6702D7CC237E@mse17be1.mse17.exchange.ms> <191c3a030908271007k6523b39fqc4b32fbf335f521b@mail.gmail.com> Message-ID: <0b1b01ca27a4$20eddde0$62c999a0$@com> I would like to comment on this statement: ?3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc.? I see this very often in this mailing-list. Would it be a good idea to create wiki page how to fine tune Freeswitch to handle as many calls as possible? Maybe such page already exist? (I?m not able to find it). If not ? I would like to help creating it. After few months when we will start developing our new product ? I will come here and will cry for help (I will be one of these 9.9/10) to tune FS to handle as many calls as possible, because that will be my goal. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 2009 m. rugpj??io 27 d. 20:07 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch I think there are 3 main problems. 1) The core developers should not be blowing their own horn about performance results in any official capacity because it will appear skewed to skeptics. 2) The core developers are too busy to bother with gathering the stats. 3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc. It sort of like if someone offered FREE CARS, take one and drive away and do whatever you want with it. People show up and say, "how fast can it go?" "how well does it handle on curves?" .. I don't know it's a FREE CAR, go drive it and you tell me. I think some people forget this is an open source project and it's up to the community to decide for themselves how well it works. I would love to find a way to save us all the time of dealing with people who try to load test first and try real calls later. The industry standard for calls is 50cps which is assuming a typical load of 25cps burst-able to 50 to cover a fault in a cluster. Most people doing 50cps all day long should be making so much money they can either afford nicer boxes or more expensive equipment if they need to scale more. FS can handle this load easily and many people tune their box to do triple that if not even more but that is all subjective to their hardware choices etc. sigh, On Thu, Aug 27, 2009 at 11:32 AM, Michael Giagnocavo wrote: Can?t we do what some vendors do? Pick the simplest config for a simple scenario, like statically bridging two channels, then publish those numbers? Or say, use one of the standard SIPP scenarios with no RTP? And so on. From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, August 27, 2009 8:44 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _______________________________________________ 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 -- 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090828/10191833/attachment-0001.html From krice at freeswitch.org Fri Aug 28 08:12:50 2009 From: krice at freeswitch.org (Ken Rice) Date: Fri, 28 Aug 2009 10:12:50 -0500 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: <0b1b01ca27a4$20eddde0$62c999a0$@com> Message-ID: The problem here is how you tune for the most calls possible is different for different objectives... There is no 1 size fits all solutions From: Mindaugas Kezys Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Fri, 28 Aug 2009 08:55:20 +0300 To: Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch I would like to comment on this statement: "3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc." ?I see this very often in this mailing-list. Would it be a good idea to create wiki page how to fine tune Freeswitch to handle as many calls as possible? Maybe such page already exist? (I'm not able to find it). If not - I would like to help creating it. After few months when we will start developing our new product - I will come here and will cry for help (I will be one of these 9.9/10) to tune FS to handle as many calls as possible, because that will be my goal. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 2009 m. rugpj??io 27 d. 20:07 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch I think there are 3 main problems. 1) The core developers should not be blowing their own horn about performance results in any official capacity because it will appear skewed to skeptics. 2) The core developers are too busy to bother with gathering the stats. 3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc. It sort of like if someone offered FREE CARS, take one and drive away and do whatever you want with it. People show up and say, "how fast can it go?" "how well does it handle on curves?" .. I don't know it's a FREE CAR, go drive it and you tell me. I think some people forget this is an open source project and it's up to the community to decide for themselves how well it works. I would love to find a way to save us all the time of dealing with people who try to load test first and try real calls later. The industry standard for calls is 50cps which is assuming a typical load of 25cps burst-able to 50 to cover a fault in a cluster. Most people doing 50cps all day long should be making so much money they can either afford nicer boxes or more expensive equipment if they need to scale more. FS can handle this load easily and many people tune their box to do triple that if not even more but that is all subjective to their hardware choices etc. sigh, On Thu, Aug 27, 2009 at 11:32 AM, Michael Giagnocavo wrote: Can't we do what some vendors do? Pick the simplest config for a simple scenario, like statically bridging two channels, then publish those numbers? Or say, use one of the standard SIPP scenarios with no RTP? And so on. From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, August 27, 2009 8:44 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. > > > Also, when that number is exceeded, how can Freeswitch server be distributed > to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _______________________________________________ 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 -- 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ 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/20090828/b6c38eff/attachment.html From janvb at live.com Fri Aug 28 14:54:35 2009 From: janvb at live.com (Jan Berger) Date: Fri, 28 Aug 2009 23:54:35 +0200 Subject: [Freeswitch-dev] mtp In-Reply-To: References: <0b1b01ca27a4$20eddde0$62c999a0$@com> Message-ID: hi, Does anyone have or know of a mtp2/3 source that can be used for testing of M2UA? Jan _________________________________________________________________ Share your memories online with anyone you want. http://www.microsoft.com/middleeast/windows/windowslive/products/photos-share.aspx?tab=1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090828/26d01049/attachment.html From andremendes2004 at gmail.com Fri Aug 28 15:20:39 2009 From: andremendes2004 at gmail.com (Andre Mendes .) Date: Fri, 28 Aug 2009 19:20:39 -0300 Subject: [Freeswitch-dev] FS + G729 Message-ID: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> Hi, I want to use G729 like Asterisk G729 in Freeswitch, any source to make this codec work with FS? I make a install asterisk with G729, but I have a problem: FS - end point 1000 - G711 make a call to end point 1001 - G729 use a asterisk trunk to make a transcoding codec. Not work yet. Please help to make this only in FS or using a Asterisk to make a transcoding. Andr? Mendes -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090828/1aa9708c/attachment.html From brian at freeswitch.org Fri Aug 28 16:35:20 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 28 Aug 2009 18:35:20 -0500 Subject: [Freeswitch-dev] FS + G729 In-Reply-To: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> References: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> Message-ID: <68890DFE-69C1-45D7-8C81-C03E3A3ED33B@freeswitch.org> We as a project will have a codec soon. Using anything else from any other vendor doesn't support the project. So please hang tight we'll have the solution soon.. /b On Aug 28, 2009, at 5:20 PM, Andre Mendes . wrote: > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to > make this codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use > a asterisk trunk to make a transcoding codec. Not work yet. Please > help to make this only in FS or using a Asterisk to make a > transcoding. > > Andr? Mendes From msc at freeswitch.org Fri Aug 28 16:36:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 28 Aug 2009 16:36:12 -0700 Subject: [Freeswitch-dev] FS + G729 In-Reply-To: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> References: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> Message-ID: <87f2f3b90908281636g29694d6du8aa6bd2947e1f8bb@mail.gmail.com> On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . wrote: > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to make > this codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use a > asterisk trunk to make a transcoding codec. Not work yet. Please help to > make this only in FS or using a Asterisk to make a transcoding. > > Andr? Mendes FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass media mode. However, no transcoding yet. It's in the works. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090828/a0503b56/attachment.html From mitul at enterux.com Fri Aug 28 20:05:13 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 29 Aug 2009 08:35:13 +0530 Subject: [Freeswitch-dev] FS + G729 In-Reply-To: <87f2f3b90908281636g29694d6du8aa6bd2947e1f8bb@mail.gmail.com> References: <9262faf90908281520i52156ef4t9a1d9d382b4452c5@mail.gmail.com> <87f2f3b90908281636g29694d6du8aa6bd2947e1f8bb@mail.gmail.com> Message-ID: Andres, We do have a solution to run g729 over freeswitch, how many channel you want ? Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 29-Aug-2009, at 5:06 AM, Michael Collins wrote: > > > On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . > wrote: > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to > make this codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use > a asterisk trunk to make a transcoding codec. Not work yet. Please > help to make this only in FS or using a Asterisk to make a > transcoding. > > Andr? Mendes > > FreeSWITCH can do G729 is "passthrough" or proxy mode as well as > bypass media mode. However, no transcoding yet. It's in the works. > -MC > > _______________________________________________ > 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/20090829/23d17bc5/attachment.html From mkezys at gmail.com Sat Aug 29 00:32:32 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Sat, 29 Aug 2009 10:32:32 +0300 Subject: [Freeswitch-dev] Scalabilty of Freeswitch In-Reply-To: References: <0b1b01ca27a4$20eddde0$62c999a0$@com> Message-ID: <0e7801ca287a$df8cd3b0$9ea67b10$@com> I can?t imagine there are unlimited cases to make this. There should be some kind of limited number of steps which would lead to that direction. Reading all about this for several years seems it is some kind of voodoo magic which is kept in secret. >From my experience it should be easy to document this and make some kind of guide for others to follow. >From good atmosphere in this mailing list I think there will be no problem for me to learn these secrets. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: 2009 m. rugpj??io 28 d. 18:13 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch The problem here is how you tune for the most calls possible is different for different objectives... There is no 1 size fits all solutions _____ From: Mindaugas Kezys Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Fri, 28 Aug 2009 08:55:20 +0300 To: Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch I would like to comment on this statement: ?3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc.? I see this very often in this mailing-list. Would it be a good idea to create wiki page how to fine tune Freeswitch to handle as many calls as possible? Maybe such page already exist? (I?m not able to find it). If not ? I would like to help creating it. After few months when we will start developing our new product ? I will come here and will cry for help (I will be one of these 9.9/10) to tune FS to handle as many calls as possible, because that will be my goal. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 2009 m. rugpj??io 27 d. 20:07 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch I think there are 3 main problems. 1) The core developers should not be blowing their own horn about performance results in any official capacity because it will appear skewed to skeptics. 2) The core developers are too busy to bother with gathering the stats. 3) 9.9/10 newcomers who try to do load testing get it wrong somewhere and we spend a month dealing with them on the list etc. It sort of like if someone offered FREE CARS, take one and drive away and do whatever you want with it. People show up and say, "how fast can it go?" "how well does it handle on curves?" .. I don't know it's a FREE CAR, go drive it and you tell me. I think some people forget this is an open source project and it's up to the community to decide for themselves how well it works. I would love to find a way to save us all the time of dealing with people who try to load test first and try real calls later. The industry standard for calls is 50cps which is assuming a typical load of 25cps burst-able to 50 to cover a fault in a cluster. Most people doing 50cps all day long should be making so much money they can either afford nicer boxes or more expensive equipment if they need to scale more. FS can handle this load easily and many people tune their box to do triple that if not even more but that is all subjective to their hardware choices etc. sigh, On Thu, Aug 27, 2009 at 11:32 AM, Michael Giagnocavo wrote: Can?t we do what some vendors do? Pick the simplest config for a simple scenario, like statically bridging two channels, then publish those numbers? Or say, use one of the standard SIPP scenarios with no RTP? And so on. From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Thursday, August 27, 2009 8:44 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch if you have the time and ability to create such a multi-dementional array of tjat data in some usable form that would be fine, it is a wiki after all. I however tend to think that there are too many variables to reliably provide any sort of real data and that the time required to do so would be quite a lot. Mike On Aug 27, 2009, at 1:37 AM, Mindaugas Kezys wrote: Maybe it would be a good idea to create wiki page just to put such kind of information in a table: Computer specs | Other comments | Codecs used | With/Without Media | Max sim. calls reached | etc That way interested persons could get a grasp what is really all about. This is very common question based on which many people measure switch capabilities, so in my opinion should be treated with that in mind (as marketing oportunity) Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: 2009 m. rugpj??io 26 d. 23:36 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Scalabilty of Freeswitch This question sounds eerily familiar... On Wed, Aug 26, 2009 at 1:15 PM, Jerry Richards wrote: Hello All, Does anyone know what the capacity of a stand-alone Freeswitch (assuming a top-of-the-line server), in terms of how many users? You can have many hundreds of users, but there are a lot of factors: network infrastructure, call volume, etc. Also, when that number is exceeded, how can Freeswitch server be distributed to accommodate a larger installation? Yes there are strategies. You definitely want a professional to assist if this is a serious production environment. There are members of the FS community who do this sort of thing, or you could email consulting at freeswitch.org to get assistance from the core FS developers. -MC _______________________________________________ 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 -- 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 iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _____ _______________________________________________ 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/20090829/4a1e73f4/attachment-0001.html From jalsot at gmail.com Sat Aug 29 01:50:41 2009 From: jalsot at gmail.com (Tamas) Date: Sat, 29 Aug 2009 10:50:41 +0200 Subject: [Freeswitch-dev] Portaudio upgrade Message-ID: <4A98EBE1.2050902@gmail.com> Hello, I'm trying to upgrade the portaudio lib due to some issues with the sound card in my box (I've found issues on 2 different boxes). I'm trying to run FS on bleeding edge distro - as this is a desktop box requiring fancy things - Ubuntu Jaunty 9.04 x86_64/i386 I know that hot new ubuntu is not supported so I'm trying to work on by myself - and give back if I have anything reasonable :) As I saw on portaudio.com, there were some improvements with ALSA handling and it should handle newer cards better now (non-mmap patch): http://portaudio.com/trac/log/portaudio/trunk/src/hostapi/alsa/pa_linux_alsa.c?rev=1415 As I've seen, there is some fix for windows's wdmks stuff too. I'm not qualified to hack PA, but I can play with it a bit :) What I would need to know is, which version (and source - as there is a separate fork e.g. at audacity too) is the one the FS trunk is having and what fixes/changes had been applied to it before merge (if I have had the exact version of included PA, I could make a diff and see by myself). From FS svn log I can see what changes were since the version bump- From FS svn log I see, that at r8835 we had the last version bump: r8835 | anthm | 2008-06-25 00:15:50 +0200 (sze, 25 j?n 2008) | 1 line merge in newer portaudio Are the changes in FS regarding PA sent to upstream? (I know, PA is used to be pretty non-responsive) Maybe it would be good to store the last PA version and source data in tree to be able to upgrade easier in the future. We are interested in improving the quality (mainly on windows) as we try to use this excellent stuff as a softphone base :) Regards, Tamas From gmaruzz at celliax.org Sat Aug 29 02:21:49 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 29 Aug 2009 11:21:49 +0200 Subject: [Freeswitch-dev] Portaudio upgrade In-Reply-To: <4A98EBE1.2050902@gmail.com> References: <4A98EBE1.2050902@gmail.com> Message-ID: <7b197bef0908290221s2aa12b05s5383d9b7d731b866@mail.gmail.com> As Walter Sobchak used to say (http://www.imdb.com/title/tt0118715/quotes): You're entering a world of pain, son. ;-) I don't know the svn revision of PA used by FS (/usr/src/freeswitch/libs/portaudio), but you can just make a diff between the one used by FS and the bleeding edge PA svn. Will probably be little diffs, you can integrate. Be aware that mod_portaudio use its own version of pablio mechanism (found in /usr/src/freeswitch/src/mod/endpoints/mod_portaudio), maybe it will works also if you integrate the changes in the PA lib (/usr/src/freeswitch/libs/portaudio). If any problem arises, its because some modification in the PA lib conflicts with the pablio mechanism. OTH, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 2009/8/29 Tamas : > Hello, > > I'm trying to upgrade the portaudio lib due to some issues with the > sound card in my box (I've found issues on 2 different boxes). > I'm trying to run FS on bleeding edge distro - as this is a desktop box > requiring fancy things - Ubuntu Jaunty 9.04 x86_64/i386 > I know that hot new ubuntu is not supported so I'm trying to work on by > myself - and give back if I have anything reasonable :) > > As I saw on portaudio.com, there were some improvements with ALSA > handling and it should handle newer cards better now (non-mmap patch): > http://portaudio.com/trac/log/portaudio/trunk/src/hostapi/alsa/pa_linux_alsa.c?rev=1415 > > As I've seen, there is some fix for windows's wdmks stuff too. I'm not > qualified to hack PA, but I can play with it a bit :) > > What I would need to know is, which version (and source - as there is a > separate fork e.g. at audacity too) is the one the FS trunk is having > and what fixes/changes had been applied to it before merge (if I have > had the exact version of included PA, I could make a diff and see by > myself). From FS svn log I can see what changes were since the version bump- > > ?From FS svn log I see, that at r8835 we had the last version bump: > > r8835 | anthm | 2008-06-25 00:15:50 +0200 (sze, 25 j?n 2008) | 1 line > merge in newer portaudio > > Are the changes in FS regarding PA sent to upstream? (I know, PA is used > to be pretty non-responsive) > Maybe it would be good to store the last PA version and source data in > tree to be able to upgrade easier in the future. > > We are interested in improving the quality (mainly on windows) as we try > to use this excellent stuff as a softphone base :) > > Regards, > ? ?Tamas > > > > _______________________________________________ > 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 mike at jerris.com Sat Aug 29 05:14:32 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 29 Aug 2009 08:14:32 -0400 Subject: [Freeswitch-dev] Portaudio upgrade In-Reply-To: <4A98EBE1.2050902@gmail.com> References: <4A98EBE1.2050902@gmail.com> Message-ID: <1D482E26-B9DC-49DF-BD22-F3E74969A35B@jerris.com> I THINK we were on the last stable snapshot at the time http://www.portaudio.com/archives/pa_stable_v19_061121.tar.gz . I don;t think anyone has taken the time to push any patches back up. Some of them for sure are hack types that would not be accepted. Others should be merged back, and I would need to see the diff to say more which are which. Mike On Aug 29, 2009, at 4:50 AM, Tamas wrote: > Hello, > > I'm trying to upgrade the portaudio lib due to some issues with the > sound card in my box (I've found issues on 2 different boxes). > I'm trying to run FS on bleeding edge distro - as this is a desktop > box > requiring fancy things - Ubuntu Jaunty 9.04 x86_64/i386 > I know that hot new ubuntu is not supported so I'm trying to work on > by > myself - and give back if I have anything reasonable :) > > As I saw on portaudio.com, there were some improvements with ALSA > handling and it should handle newer cards better now (non-mmap patch): > http://portaudio.com/trac/log/portaudio/trunk/src/hostapi/alsa/pa_linux_alsa.c?rev=1415 > > As I've seen, there is some fix for windows's wdmks stuff too. I'm not > qualified to hack PA, but I can play with it a bit :) > > What I would need to know is, which version (and source - as there > is a > separate fork e.g. at audacity too) is the one the FS trunk is having > and what fixes/changes had been applied to it before merge (if I have > had the exact version of included PA, I could make a diff and see by > myself). From FS svn log I can see what changes were since the > version bump- > > From FS svn log I see, that at r8835 we had the last version bump: > > r8835 | anthm | 2008-06-25 00:15:50 +0200 (sze, 25 j?n 2008) | 1 line > merge in newer portaudio > > Are the changes in FS regarding PA sent to upstream? (I know, PA is > used > to be pretty non-responsive) > Maybe it would be good to store the last PA version and source data in > tree to be able to upgrade easier in the future. > > We are interested in improving the quality (mainly on windows) as we > try > to use this excellent stuff as a softphone base :) > > Regards, > Tamas > > > > _______________________________________________ > 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 anthony.minessale at gmail.com Sat Aug 29 08:54:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 29 Aug 2009 10:54:22 -0500 Subject: [Freeswitch-dev] Portaudio upgrade In-Reply-To: <1D482E26-B9DC-49DF-BD22-F3E74969A35B@jerris.com> References: <4A98EBE1.2050902@gmail.com> <1D482E26-B9DC-49DF-BD22-F3E74969A35B@jerris.com> Message-ID: <191c3a030908290854m604dda6fo1a2686ac5066f137@mail.gmail.com> We don't even have any local mods to the pa lib, You should be able to drop in any newer version. They abandoned that pablio thing and we took it from the trash and hacked it up so that's part of the mod now not the lib. The last time I tried a newer svn snapshot of pa it crashed a lot so good luck maybe if they claim its stable now it will work. On Aug 29, 2009 7:19 AM, "Michael Jerris" wrote: I THINK we were on the last stable snapshot at the time http://www.portaudio.com/archives/pa_stable_v19_061121.tar.gz . I don;t think anyone has taken the time to push any patches back up. Some of them for sure are hack types that would not be accepted. Others should be merged back, and I would need to see the diff to say more which are which. Mike On Aug 29, 2009, at 4:50 AM, Tamas wrote: > Hello, > > I'm trying to upgrade the portaudio lib due... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090829/c26f6bc2/attachment.html From mike at jerris.com Sat Aug 29 10:41:01 2009 From: mike at jerris.com (Michael Jerris) Date: Sat, 29 Aug 2009 13:41:01 -0400 Subject: [Freeswitch-dev] Portaudio upgrade In-Reply-To: <191c3a030908290854m604dda6fo1a2686ac5066f137@mail.gmail.com> References: <4A98EBE1.2050902@gmail.com> <1D482E26-B9DC-49DF-BD22-F3E74969A35B@jerris.com> <191c3a030908290854m604dda6fo1a2686ac5066f137@mail.gmail.com> Message-ID: <7617AAE3-121D-4C26-9359-1E6270293BC5@jerris.com> We don't have any real code changes but there are quite a few changes for portability. (both build system and in code) On Aug 29, 2009, at 11:54 AM, Anthony Minessale wrote: > We don't even have any local mods to the pa lib, > You should be able to drop in any newer version. > They abandoned that pablio thing and we took it from the trash and > hacked it up so that's part of the mod now not the lib. > The last time I tried a newer svn snapshot of pa it crashed a lot so > good luck maybe if they claim its stable now it will work. > > >> On Aug 29, 2009 7:19 AM, "Michael Jerris" wrote: >> >> I THINK we were on the last stable snapshot at the time http://www.portaudio.com/archives/pa_stable_v19_061121.tar.gz >> . I don;t think anyone has taken the time to push any patches back >> up. Some of them for sure are hack types that would not be accepted. >> Others should be merged back, and I would need to see the diff to say >> more which are which. >> >> Mike >> On Aug 29, 2009, at 4:50 AM, Tamas wrote: > Hello, > > I'm trying >> to upgrade the portaudio lib due... >> > > _______________________________________________ > 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/20090829/51c1105e/attachment.html From erwin.davis at gmail.com Sun Aug 30 18:00:33 2009 From: erwin.davis at gmail.com (Erwin Huang) Date: Sun, 30 Aug 2009 18:00:33 -0700 Subject: [Freeswitch-dev] video support in FS Message-ID: Hi, I wonder if there is any sample application for mod_fsv. What video codec is supported? Thanks, Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20090830/e2191b04/attachment.html From anatoliy at kounitskiy.com Mon Aug 31 02:15:10 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Mon, 31 Aug 2009 12:15:10 +0300 Subject: [Freeswitch-dev] Question about mod_limit Message-ID: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com> I have question about mod_limit: is there a reason to have difference between the api applications and the dialplan? I mean when you use db function from the cli you can use insert/select/delete but when you use it from the dialplan you can only use insert/delete. #define DB_USAGE "[insert|delete]///" #define DB_DESC "save data" SWITCH_STANDARD_APP(db_function) but in the SWITCH_STANDARD_API(db_api_function) i have three options insert,delete and select What i try to accomplish is to be able to insert values from the dialplan and after that to be able to use them ( to select and use the result) from another extension. Thank you in advanced, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From anatoliy at kounitskiy.com Mon Aug 31 07:32:05 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Mon, 31 Aug 2009 17:32:05 +0300 Subject: [Freeswitch-dev] Question about mod_limit In-Reply-To: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com> References: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com> Message-ID: <1cd828b60908310732p1c6db054kf3db1ea842e5a004@mail.gmail.com> OK, I was able to answer one of the questions: if I want to use the function db - i can insert and delete: but if i want to select i have to use the API function db: Probably there is a reason, that the dialplan application db can't do select - someone input here? I was looking something like this: but at the end i found a solution. I'm just searching for a reason :) Bear in mind that I'm newbie in the FreeSwitch :) On Mon, Aug 31, 2009 at 12:15 PM, Anatoliy Kounitskiy wrote: > I have question about mod_limit: is there a reason to have difference > between the api applications and the dialplan? I mean when you use db > function from the cli you can use insert/select/delete but when you > use it from the dialplan you can only use insert/delete. > > #define DB_USAGE "[insert|delete]///" > #define DB_DESC "save data" > > SWITCH_STANDARD_APP(db_function) > > but in the > SWITCH_STANDARD_API(db_api_function) > i have three options insert,delete and select > > What i try to accomplish is to be able to insert values from the > dialplan and after that to be able to use them ( to select and use the > result) from another extension. > > Thank you in advanced, > -- > Anatoliy Kounitskiy > ------------------------- > E-mail: anatoliy at kounitskiy.com > Mobile: +359898913540 > -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From jmesquita at freeswitch.org Sun Aug 30 22:09:35 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 31 Aug 2009 02:09:35 -0300 Subject: [Freeswitch-dev] video support in FS In-Reply-To: References: Message-ID: If I am not mistaken, mod_fsv only replay RTP packets, it does not do any encoding/decoding at all. jmesquita On Sun, Aug 30, 2009 at 10:00 PM, Erwin Huang wrote: > Hi, > > I wonder if there is any sample application for mod_fsv. What video codec > is supported? Thanks, > > Regards, > > > > _______________________________________________ > 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/20090831/e668a4fc/attachment-0001.html From mauro at nohad.com.br Mon Aug 31 08:49:51 2009 From: mauro at nohad.com.br (mauro at nohad.com.br) Date: Mon, 31 Aug 2009 12:49:51 -0300 Subject: [Freeswitch-dev] Fwd: FS + G729 Message-ID: <003f01ca2a52$ada384a0$08ea8de0$@com.br> Hi there Interesting point. We have licenses for 723 and 729 for asterisk, and we also need to use asterisk distribution to talk with our voice card for PSTN. We don?t have the drives for this card developed for freeswitch ?yet?. So, for us the freeswitch is a great solution, and we?re beginning to use it, but ?for now? we also must use asterisk ?for some things? that: or we don?t know how to do yet, or we feel more ?comfortable? with, or we ?have to?. I think that this is a conservative and secure approach for us, since we just beginning ?to scratch? freeswitch. We tried to do exactly what is described in the message (bottom) but we have no success. Like I said we ?use and must to use? asterisk for now, and we already have the 729 and 723 licenses. I know that 729 (and 723?) license like asterisk is on the way, and also know that there is at least one vendor for 729 for freeswitch, but it will be great if someone could give us the direction to make a extension registered in freeswitch to talk with another thought asterisk to make the codec ?trans-coding? in asterisk. I also support the idea to solve things like this in the freeswitch project (we?re not talking here about asterisk) but ?for now? (for us) to make a good use of the 923459283752938 good things in freeswitch we must to use it together with asterisk. Any idea to this direction (transcoding thought asterisk) would be great appreciated. Regards, Mauro ---------- Forwarded message ---------- From: Michael Collins < msc at freeswitch.org> Date: Fri, Aug 28, 2009 at 8:36 PM Subject: Re: [Freeswitch-dev] FS + G729 To: freeswitch-dev at lists.freeswitch.org On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . < andremendes2004 at gmail.com> wrote: Hi, I want to use G729 like Asterisk G729 in Freeswitch, any source to make this codec work with FS? I make a install asterisk with G729, but I have a problem: FS - end point 1000 - G711 make a call to end point 1001 - G729 use a asterisk trunk to make a transcoding codec. Not work yet. Please help to make this only in FS or using a Asterisk to make a transcoding. Andr? Mendes FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass media mode. However, no transcoding yet. It's in the works. -MC _______________________________________________ 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/20090831/edea74d1/attachment-0001.html From jmesquita at freeswitch.org Mon Aug 31 11:55:53 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 31 Aug 2009 15:55:53 -0300 Subject: [Freeswitch-dev] Fwd: FS + G729 In-Reply-To: <003f01ca2a52$ada384a0$08ea8de0$@com.br> References: <003f01ca2a52$ada384a0$08ea8de0$@com.br> Message-ID: Mauro, I am sorry for the curiosity and offtopic question, but what boards are you talking about? Regards, jmesquita On Mon, Aug 31, 2009 at 12:49 PM, wrote: > Hi there > > Interesting point. We have licenses for 723 and 729 for asterisk, and we > also need to use asterisk distribution to talk with our voice card for PSTN. > We don?t have the drives for this card developed for freeswitch ?yet?. > > So, for us the freeswitch is a great solution, and we?re beginning to use > it, but ?for now? we also must use asterisk ?for some things? that: or we > don?t know how to do yet, or we feel more ?comfortable? with, or we ?have > to?. I think that this is a conservative and secure approach for us, since > we just beginning ?to scratch? freeswitch. > > We tried to do exactly what is described in the message (bottom) but we > have no success. Like I said we ?use and must to use? asterisk for now, and > we already have the 729 and 723 licenses. I know that 729 (and 723?) license > like asterisk is on the way, and also know that there is at least one vendor > for 729 for freeswitch, but it will be great if someone could give us the > direction to make a extension registered in freeswitch to talk with > another thought asterisk to make the codec ?trans-coding? in asterisk. > > I also support the idea to solve things like this in the freeswitchproject (we?re not talking here about asterisk) but ?for now? (for us) to > make a good use of the 923459283752938 good things in freeswitch we must > to use it together with asterisk. > > Any idea to this direction (transcoding thought asterisk) would be great > appreciated. > > Regards, > > Mauro > > ---------- Forwarded message ---------- > From: *Michael Collins* > Date: Fri, Aug 28, 2009 at 8:36 PM > Subject: Re: [Freeswitch-dev] FS + G729 > To: freeswitch-dev at lists.freeswitch.org > > > > On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . < > andremendes2004 at gmail.com> wrote: > > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to make > this codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use a > asterisk trunk to make a transcoding codec. Not work yet. Please help to > make this only in FS or using a Asterisk to make a transcoding. > > Andr? Mendes > > > FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass > media mode. However, no transcoding yet. It's in the works. > -MC > > > > _______________________________________________ > 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/20090831/9aaf69bd/attachment.html From krice at freeswitch.org Mon Aug 31 12:00:58 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 31 Aug 2009 14:00:58 -0500 Subject: [Freeswitch-dev] Fwd: FS + G729 In-Reply-To: Message-ID: If your requirements are in the 100 channels or less range the TC400 works in freeswitch From: Jo?o Mesquita Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Mon, 31 Aug 2009 15:55:53 -0300 To: "freeswitch-dev at lists.freeswitch.org" Subject: Re: [Freeswitch-dev] Fwd: FS + G729 Mauro, I am sorry for the curiosity and offtopic question, but what boards are you talking about? Regards, jmesquita On Mon, Aug 31, 2009 at 12:49 PM, wrote: > Hi there > > Interesting point. We have licenses for 723 and 729 for asterisk, and we also > need to use asterisk distribution to talk with our voice card for PSTN. We > don?t have the drives for this card developed for freeswitch ?yet?. > > So, for us the freeswitch is a great solution, and we?re beginning to use it, > but ?for now? we also must use asterisk ?for some things? that: or we don?t > know how to do yet, or we feel more ?comfortable? with, or we ?have to?. I > think that this is a conservative and secure approach for us, since we just > beginning ?to scratch? freeswitch. > > We tried to do exactly what is described in the message (bottom) but we have > no success. Like I said we ?use and must to use? asterisk for now, and we > already have the 729 and 723 licenses. I know that 729 (and 723?) license like > asterisk is on the way, and also know that there is at least one vendor for > 729 for freeswitch, but it will be great if someone could give us the > direction to make a extension registered in freeswitch to talk with another > thought asterisk to make the codec ?trans-coding? in asterisk. > > I also support the idea to solve things like this in the freeswitch project > (we?re not talking here about asterisk) but ?for now? (for us) to make a good > use of the 923459283752938 good things in freeswitch we must to use it > together with asterisk. > > Any idea to this direction (transcoding thought asterisk) would be great > appreciated. > > Regards, > > Mauro > > ---------- Forwarded message ---------- > From: Michael Collins > > Date: Fri, Aug 28, 2009 at 8:36 PM > Subject: Re: [Freeswitch-dev] FS + G729 > To: freeswitch-dev at lists.freeswitch.org > > > > > > On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . > wrote: > > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to make this > codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use a asterisk > trunk to make a transcoding codec. Not work yet. Please help to make this only > in FS or using a Asterisk to make a transcoding. > > Andr? Mendes > > > FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass media > mode. However, no transcoding yet. It's in the works. > -MC > > > > _______________________________________________ > 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/20090831/88c7c523/attachment.html From mitul at enterux.com Mon Aug 31 12:06:38 2009 From: mitul at enterux.com (Mitul Limbani) Date: Tue, 1 Sep 2009 00:36:38 +0530 Subject: [Freeswitch-dev] Fwd: FS + G729 In-Reply-To: References: <003f01ca2a52$ada384a0$08ea8de0$@com.br> Message-ID: Adding to the curiosity, rather ending it. Is it TC400B card from Digium? Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 01-Sep-2009, at 12:25 AM, Jo?o Mesquita wrote: > Mauro, > > I am sorry for the curiosity and offtopic question, but what boards > are you talking about? > > Regards, > > jmesquita > > On Mon, Aug 31, 2009 at 12:49 PM, wrote: > Hi there > > Interesting point. We have licenses for 723 and 729 for asterisk, > and we also need to use asterisk distribution to talk with our voice > card for PSTN. We don?t have the drives for this card developed for > freeswitch ?yet?. > > So, for us the freeswitch is a great solution, and we?re beginning t > o use it, but ?for now? we also must use asterisk ?for some > things? that: or we don?t know how to do yet, or we feel more > ?comfortable? with, or we ?have to?. I think that this is a > conservative and secure approach for us, since we just beginning ?to > scratch? freeswitch. > > We tried to do exactly what is described in the message (bottom) but > we have no success. Like I said we ?use and must to use? asterisk > for now, and we already have the 729 and 723 licenses. I know that 7 > 29 (and 723?) license like asterisk is on the way, and also know tha > t there is at least one vendor for 729 for freeswitch, but it will b > e great if someone could give us the direction to make a extension r > egistered in freeswitch to talk with another thought asterisk to mak > e the codec ?trans-coding? in asterisk. > > I also support the idea to solve things like this in the freeswitch > project (we?re not talking here about asterisk) but ?for > now? (for us) to make a good use of the 923459283752938 good things > in freeswitch we must to use it together with asterisk. > > Any idea to this direction (transcoding thought asterisk) would be > great appreciated. > > Regards, > > Mauro > > ---------- Forwarded message ---------- > From: Michael Collins > Date: Fri, Aug 28, 2009 at 8:36 PM > Subject: Re: [Freeswitch-dev] FS + G729 > To: freeswitch-dev at lists.freeswitch.org > > > > > On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . > wrote: > > Hi, > I want to use G729 like Asterisk G729 in Freeswitch, any source to > make this codec work with FS? > > I make a install asterisk with G729, but I have a problem: > > FS - end point 1000 - G711 make a call to end point 1001 - G729 use > a asterisk trunk to make a transcoding codec. Not work yet. Please > help to make this only in FS or using a Asterisk to make a > transcoding. > > Andr? Mendes > > > FreeSWITCH can do G729 is "passthrough" or proxy mode as well as > bypass media mode. However, no transcoding yet. It's in the works. > -MC > > > > _______________________________________________ > 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/20090901/338449ac/attachment-0001.html From krice at freeswitch.org Mon Aug 31 12:12:34 2009 From: krice at freeswitch.org (Ken Rice) Date: Mon, 31 Aug 2009 14:12:34 -0500 Subject: [Freeswitch-dev] Fwd: FS + G729 In-Reply-To: Message-ID: Yes its their transcoding card... Rumor has it there are other cards that will be available soon to do the same thing... And I know for a fact that the Core FS team are working diligently to get the official FS G729 software module out From: Mitul Limbani Reply-To: "freeswitch-dev at lists.freeswitch.org" Date: Tue, 1 Sep 2009 00:36:38 +0530 To: "freeswitch-dev at lists.freeswitch.org" Subject: Re: [Freeswitch-dev] Fwd: FS + G729 Adding to the curiosity, rather ending it. Is it TC400B card from Digium? Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 01-Sep-2009, at 12:25 AM, Jo?o Mesquita wrote: > Mauro, > > I am sorry for the curiosity and offtopic question, but what boards are you > talking about? > > Regards, > > jmesquita > > On Mon, Aug 31, 2009 at 12:49 PM, > wrote: >> Hi there >> >> Interesting point. We have licenses for 723 and 729 for asterisk, and we also >> need to use asterisk distribution to talk with our voice card for PSTN. We >> don?t have the drives for this card developed for freeswitch ?yet?. >> >> So, for us the freeswitch is a great solution, and we?re beginning to use it, >> but ?for now? we also must use asterisk ?for some things? that: or we don?t >> know how to do yet, or we feel more ?comfortable? with, or we ?have to?. I >> think that this is a conservative and secure approach for us, since we just >> beginning ?to scratch? freeswitch. >> >> We tried to do exactly what is described in the message (bottom) but we have >> no success. Like I said we ?use and must to use? asterisk for now, and we >> already have the 729 and 723 licenses. I know that 729 (and 723?) license >> like asterisk is on the way, and also know that there is at least one vendor >> for 729 for freeswitch, but it will be great if someone could give us the >> direction to make a extension registered in freeswitch to talk with another >> thought asterisk to make the codec ?trans-coding? in asterisk. >> >> I also support the idea to solve things like this in the freeswitch project >> (we?re not talking here about asterisk) but ?for now? (for us) to make a good >> use of the 923459283752938 good things in freeswitch we must to use it >> together with asterisk. >> >> Any idea to this direction (transcoding thought asterisk) would be great >> appreciated. >> >> Regards, >> >> Mauro >> >> ---------- Forwarded message ---------- >> From: Michael Collins > >> Date: Fri, Aug 28, 2009 at 8:36 PM >> Subject: Re: [Freeswitch-dev] FS + G729 >> To: freeswitch-dev at lists.freeswitch.org >> >> >> >> >> >> On Fri, Aug 28, 2009 at 3:20 PM, Andre Mendes . > > wrote: >> >> Hi, >> I want to use G729 like Asterisk G729 in Freeswitch, any source to make this >> codec work with FS? >> >> I make a install asterisk with G729, but I have a problem: >> >> FS - end point 1000 - G711 make a call to end point 1001 - G729 use a >> asterisk trunk to make a transcoding codec. Not work yet. Please help to make >> this only in FS or using a Asterisk to make a transcoding. >> >> Andr? Mendes >> >> >> FreeSWITCH can do G729 is "passthrough" or proxy mode as well as bypass media >> mode. However, no transcoding yet. It's in the works. >> -MC >> >> >> >> _______________________________________________ >> 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/20090831/2cf994b1/attachment.html From erwin.davis at gmail.com Mon Aug 31 12:57:30 2009 From: erwin.davis at gmail.com (Erwin Huang) Date: Mon, 31 Aug 2009 12:57:30 -0700 Subject: [Freeswitch-dev] video support in FS In-Reply-To: References: Message-ID: thanks,Jo?o. From the http://wiki.freeswitch.org/wiki/Modules, it doesnot give any info on configuration and usage. 2009/8/30 Jo?o Mesquita > If I am not mistaken, mod_fsv only replay RTP packets, it does not do any > encoding/decoding at all. > > jmesquita > > On Sun, Aug 30, 2009 at 10:00 PM, Erwin Huang wrote: > >> Hi, >> >> I wonder if there is any sample application for mod_fsv. What video codec >> is supported? Thanks, >> >> Regards, >> >> >> >> _______________________________________________ >> 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/20090831/bf0177d5/attachment.html